Table Of Contents
Cisco IOS Voice, Video, and Fax Commands:
R Through Sh
register e164
registered-caller ring
req-qos
reset
resource threshold
response-timeout
retry-delay
retry-limit
retry (SIP user-agent)
ring
ring cadence
ring frequency
ring number
roaming (dial-peer)
roaming (settlement)
rtsp client session history duration
rtsp client session history records
rule
security
sequence-numbers
server (RLM)
server registration-port
server trigger
session
session protocol
session protocol (Voice over Frame Relay)
session protocol aal2
session protocol multicast
session target (VoATM)
session target (VoFR)
session target (VoIP)
session transport
set
settle-call
settlement
settlement roam-pattern
sgcp
sgcp call-agent
sgcp graceful-shutdown
sgcp max-waiting-delay
sgcp modem passthru
sgcp quarantine-buffer disable
sgcp request retries
sgcp request timeout
sgcp restart
sgcp retransmit timer
sgcp timer
sgcp tse payload
show aal2 profile
show atm video-voice address
show backhaul-session-manager group
show backhaul-session-manager session
show backhaul-session-manager set
show call active
show call application voice
show call fallback cache
show call fallback config
show call fallback stats
show call history
show call history video record
show call history voice record
show call resource voice stats
show call resource voice threshold
show call rsvp-sync conf
show call rsvp-sync stats
show cdapi
show ces clock-select
show connect
show controllers rs366
show controllers timeslots
show controllers voice
show csm
show dial-peer video
show dial-peer voice
show dialplan incall number
show dialplan number
show frame-relay vofr
show gatekeeper calls
show gatekeeper endpoints
show gatekeeper gw-type-prefix
show gatekeeper servers
show gatekeeper status
show gatekeeper zone prefix
show gatekeeper zone status
show gateway
show interface dspfarm
show mgcp
show mgcp connection
show mgcp endpoint
show mgcp statistics
show num-exp
show pots csm
show pots status
show proxy h323 calls
show proxy h323 detail-call
show proxy h323 status
show rawmsg
show rlm group statistics
show rlm group status
show rlm group timer
show rtsp client session
show rudpv0 failures
show rudpv0 statistics
show rudpv1
show settlement
show sgcp connection
show sgcp endpoint
show sgcp statistics
show sip-ua
show ss7 mtp2 ccb
show ss7 mtp2 state
show ss7 mtp2 stats
show ss7 mtp2 timer
show ss7 mtp2 variant
show ss7 sm session
show ss7 sm set
show ss7 sm stats
show translation-rule
show vfc
show vfc cap-list
show vfc default-file
show vfc directory
show vfc version
show video call summary
show voice busyout
show voice call
show voice dsp
show voice permanent-call
show voice port
show voice trunk-conditioning signaling
show voice trunk-conditioning supervisory
show vrm active_calls
show vrm vdevices
shut
shutdown (dial-peer)
shutdown (DS1)
shutdown (gatekeeper)
shutdown (RLM)
shutdown (settlement)
shutdown (voice-port)
Cisco IOS Voice, Video, and Fax Commands:
R Through Sh
This chapter presents the commands to configure and maintain Cisco IOS voice, video, and fax applications. The commands are presented in alphabetical order beginning with R. Some commands required for configuring voice, video, and fax may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands.
For detailed information on how to configure these applications and features, refer to the Cisco IOS Voice, Video, and Fax Configuration Guide.
register e164
To configure a gateway to register or deregister (remove the registration for) a fully qualified plain old telephone service (POTS) dial-peer E.164 address with a gatekeeper, use the register e164 command in dial-peer configuration mode. To deregister an E.164 address, use the no form of this command.
register e164
no register e164
Syntax Description
This command has no keywords or arguments.
Defaults
No E.164 addresses are registered until you enter this command.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use this command to register the E.164 address of an analog telephone line attached to a Foreign Exchange Station (FXS) port on a router. The gateway automatically registers fully qualified E164 addresses. Use the no register e164 command to deregister an address. Use the register e164 command to register a deregistered address.
Before you automatically or manually register an E.164 address with a gatekeeper, you must create a dial peer (using the dial-peer command), assign an FXS port to the peer (using the port command), and assign an E.164 address (using the destination-pattern command). The E.164 address must be a fully qualified address. For example, +5551212, 5551212, and 4085551212 are fully qualified addresses; 408555.... is not a fully qualified address. E.164 addresses are registered only for active interfaces—those that are not shut down. If an FXS port or its interface is shut down, the corresponding E.164 address is deregistered.
Tips
You can use the show gateway command to find out if the gateway is connected to a gatekeeper and if a fully qualified E.164 address is assigned to the gateway. Use the zone-prefix command at the gatekeeper to define prefix patterns, such as 408555...., that apply to one or more gateways.
Examples
The following command sequence places the gateway in dial-peer configuration mode, assigns an E.164 address to the interface, and registers that address with the gatekeeper:
destination-pattern 5551212
The following commands deregister an address with the gatekeeper:
The following example shows that you must have a connection to a gatekeeper and define a unique E.164 address before you can register an address:
ERROR-register-e164:Dial-peer destination-pattern is not a full E.164 number
ERROR-register-e164:No gatekeeper
Related Commands
Command
|
Description
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dial-peer
|
Enters dial-peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
|
port
|
Enables an interface on a PA-4R-DTR to operate as a concentrator port.
|
show gateway
|
Displays the current gateway status.
|
zone prefix
|
Configures the gatekeeper with knowledge of its own prefix and the prefix of any remote zone.
|
registered-caller ring
To configure the Nariwake service registered caller ring cadence, use the registered-caller ring command in dial-peer configuration mode.
registered-caller ring cadence
Syntax Description
cadence
|
A value of 0, 1, or 2. The default ring cadence for registered callers is 1 and for unregistered callers is 0. The on and off periods of ring 0 (normal ringing signals) and ring 1 (ringing signals for the Nariwake service) are defined in the NTT user manual.
|
Defaults
The default Nariwake service registered caller ring cadence is ring 1.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1.(2)XF
|
The command registered-caller ring was introduced on the Cisco 800 series routers.
|
Usage Guidelines
If your ISDN line is provisioned for the I Number or dial-in services, you must also configure a dial peer by using the destination-pattern not-provided command. Either port 1 or port 2 can be configured under this dial peer. The router then forwards the incoming call to voice port 1. (See the "Examples" section below.
If more than one dial peer is configured with the destination-pattern not-provided command, the router uses the first configured dial peer for the incoming calls. To display the Nariwake ring cadence setting, use the show run command.
Examples
The following example sets the ring cadence for registered callers to 2.
req-qos
To specify the desired quality of service to be used in reaching a specified dial peer, use the req-qos command in dial-peer configuration mode. To restore the default value for this command, use the no form of this command.
req-qos {best-effort | controlled-load | guaranteed-delay}
no req-qos
Syntax Description
best-effort
|
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation.
|
controlled-load
|
Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded.
|
guaranteed-delay
|
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.
|
Defaults
best-effort
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series routers.
|
Usage Guidelines
This command is applicable only to VoIP dial peers.
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
Examples
The following example configures guaranteed-delay as the desired (requested) quality of service to a dial peer:
Related Commands
Command
|
Description
|
acc-qos
|
Defines the acceptable QoS for any inbound and outbound call on a VoIP dial peer.
|
reset
To reset a set of digital signal processors (DSPs), use the reset command in global configuration mode.
reset number
Syntax Description
number
|
Specifies the number of DSPs to be reset. The number of DSPs ranges from 0 to 30.
|
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
12.0(5)XE
|
This command was introduced on the Cisco 7200 series routers.
|
12.0(7)T
|
This command was integrated into the Cisco IOS Release 12.0(7)T.
|
Examples
The following example displays the reset command configuration for DSP 1:
01:24:54:%DSPRM-5-UPDOWN: DSP 1 in slot 1, changed state to up
resource threshold
To configure a gateway to report H.323 resource availability to the its gatekeeper, use the resource threshold command in gateway configuration mode. To disable gateway resource-level reporting, use the no form of this command.
resource threshold [all] [high percentage-value] [low percentage-value]
no resource threshold
Syntax Description
all
|
(Optional) Applies the high- and low- parameter settings to all monitored H.323 resources. This is the default condition.
|
high percentage-value
|
(Optional) A resource utilization level that triggers a Resource Availability Indicator (RAI) message indicating that H.323 resource use is high. Enter a number between 1 and 100 that represents the high-resource utilization percentage. A value of 100 specifies high-resource usage when any H.323 resource is unavailable. The default is 90 percent.
|
low percentage-value
|
(Optional) Resource utilization level that triggers an RAI message indicating that H.323 resource usage has dropped below the high-usage level. Enter a number between 1 and 100 that represents the acceptable resource utilization percentage. After the gateway sends a high-utilization message, it waits to send the resource recovery message until the resource use drops below the value defined by the low parameter. The default is 90 percent.
|
Defaults
Reports low resources when 90 percent of resources are in use, and reports resource availability when resource use drops below 90 percent.
Command Modes
Gateway configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
The resource threshold command defines the resource load levels that trigger Resource Availability Indicator (RAI) messages. To view the monitored resources, enter the show gateway command.
The monitored H.323 resources include digital signal processor (DSP) channels and DS0s. Use the show call resource voice stats command to see the total amount of resources available for H.323 calls.
Note
The DS0 resources that are monitored for H.323 calls are limited to the ones that are associated with a voice POTS dial peer.
See the dial-peer configuration commands for details on how to associate a dial peer with a PRI or CAS group.
When any monitored H.323 resources exceed the threshold level defined by the high parameter, the gateway sends an RAI message to the gatekeeper with the AlmostOutOfResources field flagged. This message reports high resource usage.
When all gateway H.323 resources drop below the level defined by the low parameter, the gateway sends the RAI message to the gatekeeper with the AlmostOutOfResources field cleared.
When a gatekeeper can choose between multiple gateways for call completion, the gatekeeper uses internal priority settings and gateway resource statistics to determine which gateway to use. When all other factors are equal, a gateway that has available resources will be chosen over a gateway that has reported limited resources.
Examples
The following command defines the H.323 resource limits for a gateway:
resource threshold high 70 low 60
Related Commands
Command
|
Description
|
show call resource voice stats
|
Displays resource statistics for an H.323 gateway.
|
show call resource voice threshold
|
Displays the threshold configuration settings and status for an H.323 gateway.
|
show gateway
|
Displays the current gateway status.
|
response-timeout
To configure the maximum time to wait for a response from a server, use the response-timeout command in settlement configuration mode. To restore the default value of this command, use the no form of this command.
response-timeout number
no response-timeout number
Syntax Description
number
|
Response waiting time in seconds.
|
Defaults
The default response timeout is one (1) second.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
If no response is received within the response-timeout time limit, the current connection ends, and the router attempts to contact the next service point.
Examples
The following example illustrates a response-timeout set to 1 second.
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time for which a connection is maintained after completion of a communication exchange.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
retry-limit
|
Sets the maximum number of attempts to connect to the provider.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement mode and specifies the attributes specific to a settlement provider.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown/no shutdown
|
Deactivates the settlement provider/activates the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
url
|
Specifies the Internet service provider address.
|
retry-delay
To set the time between attempts to connect with the settlement provider, use the retry-delay command in settlement configuration mode. To restore the default value, use the no form of this command.
retry-delay number
no retry-delay
Syntax Description
number
|
Length of time (in seconds) between attempts to connect with the settlement provider. The valid range for retry delay is from 1 to 600 seconds.
|
Defaults
The default retry delay is two seconds.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 and 3600 series routers and the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
After exhausting all service points for the provider, the router is delayed for the specified length of time before resuming connection attempts.
Examples
The following example sets a retry value of 15 seconds:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time for which a connection is maintained after completion of a communication exchange.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-limit
|
Sets the maximum number of attempts to connect to the provider.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown/no shutdown
|
Deactivates the settlement provider/activates the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
retry-limit
To set the maximum number of attempts to connect to the provider, use the retry-limit command in settlement configuration mode. To restore the default value, use the no form of this command.
retry-limit number
no retry-limit number
Syntax Description
number
|
Maximum number of connection attempts in addition to the first attempt.
|
Defaults
The default retry limit is one (1) retry.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
If no connection is established after the configured retries, the router ceases connection attempts. The retry limit number does not count the initial connection attempt. A retry limit of one (default) results in a total of two connection attempts to every service point.
Examples
The following example sets the number of retries to 1:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time for which a connection is maintained after a communication exchange is complete.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
session-timeout
|
Sets the length of interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement mode and specifies the attributes specific to a settlement provider.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown
|
Brings up the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
retry (SIP user-agent)
To configure the number of retry attempts for Session Initiation Protocol (SIP) messages, use the retry command in SIP user-agent configuration mode. To reset this command to the default value, use the no form of this command.
retry {invite number | response number | bye number | cancel number}
no retry {invite number | response number | bye number | cancel number}
Syntax Description
invite number
|
Number of INVITE retries: 1 through 10 are valid inputs; default = 6.
|
response number
|
Number of RESPONSE retries: 1 through 10 are valid inputs; default = 6.
|
bye number
|
Number of BYE retries: 1 through 10 are valid inputs; default = 10.
|
cancel number
|
Number of CANCEL retries: 1 through 10 are valid inputs; default = 10.
|
Defaults
invite: 6
response: 6
bye: 10
cancel: 10
Command Modes
SIP user-agent configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
In the following example, the number of invite retries has been set to 5.
Related Commands
Command
|
Description
|
sip-ua
|
Enables the sip-ua configuration commands, with which you configure the user agent.
|
ring
To set up a distinctive ring for your connected telephones, fax machines, or modems, use the ring command in interface configuration mode. To disable the specified distinctive ring, use the no form of this command.
ring cadence-number
no ring cadence-number
Syntax Description
cadence-number
|
Number from 0 through 2:
• Type 0 is a primary ringing cadence—default ringing cadence for the country your router is in.
• Type 1 is a distinctive ring—0.8 seconds on, 0.4 seconds off, 0.8 seconds on, 0.4 seconds off.
• Type 2 is a distinctive ring—0.4 seconds on, 0.2 seconds off, 0.4 seconds on, 0.2 seconds off, 0.8 seconds on, 4 seconds off.
|
Defaults
The default is 0.
Command Modes
Interface configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco 800 series router.
|
Usage Guidelines
This command applies to Cisco 800 series routers.
You can specify this command when creating a dial peer. This command will not work if it is not specified within the context of a dial peer. For information on creating a dial peer, refer to the Cisco 800 Series Routers Software Configuration Guide.
Examples
The following example specifies the type 1 distinctive ring:
Related Commands
Command
|
Description
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dial-peer voice
|
Enters dial-peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
|
no call-waiting
|
Disables call waiting.
|
port (dial-peer)
|
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
|
pots distinctive-ring-guard-time
|
Specifies a delay in which a telephone port can be rung after a previous call is disconnected (for Cisco 800 series routers).
|
ring
|
Sets up a distinctive ring for telephones, fax machines, or modems connected to a Cisco 800 series router.
|
show dial-peer voice
|
Displays configuration information and call statistics for dial peers.
|
ring cadence
To specify the ring cadence for a Foreign Exchange Station (FXS) voice port, use the ring cadence command in voice-port configuration mode. To restore the default value, use the no form of this command.
ring cadence {pattern-number | define pulse interval}
no ring cadence
Syntax Description
pattern-number
|
Predefined ring cadence patterns. Each pattern specifies a ring-pulse time and a ring-interval time.
• pattern01—2 seconds on, 4 seconds off
• pattern02—1 second on, 4 seconds off
• pattern03—1.5 seconds on, 3.5 seconds off
• pattern04—1 second on, 2 seconds off
• pattern05—1 second on, 5 seconds off
• pattern06—1 second on, 3 seconds off
• pattern07—0.8 second on, 3.2 seconds off
• pattern08—1.5 seconds on, 3 seconds off
• pattern09—1.2 seconds on, 3.7 seconds off
• pattern09—1.2 seconds on, 4.7 seconds off
• pattern11—0.4 second on, 0.2 second off, 0.4 second on, 2 seconds off
• pattern12—0.4 second on, 0.2 second off, 0.4 second on, 2.6 seconds off
|
define
|
User-definable ring cadence pattern. Each number pair specifies one ring-pulse time and one ring-interval time. You must enter numbers in pairs, and you can enter from 1 to 6 pairs. The second number in the last pair that you enter specifies the interval between rings.
|
pulse
|
A number (1 or 2 digits) specifying ring pulse (on) time in hundreds of milliseconds.
The range is from 1 to 50, for pulses of 100 ms to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
|
interval
|
A number (1 or 2 digits) specifying ring interval (off) time in hundreds of milliseconds.
The range is from 1 to 50, for pulses of 100 to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
|
Defaults
Ring cadence defaults to the pattern you specify with the cptone command.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers, and the patternXX keyword was introduced.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
The patternXX keyword provides preset ring cadence patterns for use on any platform. The define keyword allows you to create a custom ring cadence. On the Cisco 2600 and 3600 series routers, only one or two pairs of digits can be entered under the define keyword.
Examples
The following example configures the ring cadence for 1 second on and 4 seconds off on voice port 1/1 on a Cisco MC3810 multiservice concentrator:
The following example configures the ring cadence for 1 second on, 1 second off, 1 second on, and
5 seconds off on voice port 1/2 on a Cisco MC3810 multiservice concentrator:
ring cadence define 10 10 10 50
The following example configures the ring cadence for 1 second on and 2 seconds off on voice port 1/0/0 on a Cisco 2600 or 3600 series router:
Related Commands
Command
|
Description
|
ring frequency
|
Specifies the ring frequency for a specified FXS voice port.
|
cptone
|
Specifies the default tone, ring, and cadence settings according to country.
|
ring frequency
To specify the ring frequency for a specified Foreign Exchange Station (FXS) voice port, use the ring frequency command in voice-port configuration mode. To restore the default value, use the no form of this command.
ring frequency number
no ring frequency number
Syntax Description
number
|
Ring frequency (hertz) used in the FXS interface. Valid entries on the Cisco 3600 series are 25 and 50. Valid entries on the Cisco MC3810 multiservice concentrator are 20 and 30.
|
Defaults
25 Hz on the Cisco 3600 series routers and 20 Hz on the Cisco MC3810 multiservice concentrators.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Use the ring frequency command to select a specific ring frequency for an FXS voice port. Use the no form of this command to reset the default value. The ring frequency you select must match the connected equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring frequency is usually country-dependent. You should take into account the appropriate ring frequency for your area before configuring this command.
This command does not affect ringback, which is the ringing a user hears when placing a remote call.
Examples
The following example configures the ring frequency on the Cisco 3600 series for 25 Hz:
The following example configures the ring frequency on the Cisco MC3810 multiservice concentrator for 20 Hz:
Related Commands
Command
|
Description
|
ring cadence
|
Specifies the ring cadence for an FXS voice port on the Cisco MC3810 multiservice concentrator.
|
ring number
|
Specifies the number of rings for a specified FXO voice port.
|
ring number
To specify the number of rings for a specified Foreign Exchange Office (FXO) voice port, use the ring number command in voice-port configuration mode. To restore the default value, use the no form of this command.
ring number number
no ring number number
Syntax Description
number
|
Number of rings detected before answering the call. Valid entries are numbers from 1 to 10. The default is 1.
|
Defaults
One ring
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
Usage Guidelines
Use the ring number command to set the maximum number of rings to be detected before answering a call over an FXO voice port. Use the no form of this command to reset the default value, which is one ring.
Normally, this command should be set to the default so that incoming calls are answered quickly. If you have other equipment available on the line to answer incoming calls, you might want to set the value higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if the equipment online did not answer the incoming call in the configured number of rings.
This command is not applicable to Foreign Exchange Station (FXS) or E&M interfaces because they do not receive ringing on incoming calls.
Examples
The following example on the Cisco 3600 series sets five rings as the maximum number of rings to be detected before closing a connection over this voice port:
The following example on the Cisco MC3810 multiservice concentrator sets five rings as the maximum number of rings to be detected before closing a connection over this voice port:
Related Commands
Command
|
Description
|
ring frequency
|
Specifies the ring frequency for a specified FXS voice port.
|
roaming (dial-peer)
To enable the roaming capability for the dial peer, use the roaming command in dial-peer configuration mode. To disable the roaming capability, use the no form of this command.
roaming
no roaming
Syntax Description
This command has no arguments or keywords.
Defaults
No roaming
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server`.
|
Usage Guidelines
Enable the roaming capability of a dial peer if that dial peer can terminate roaming calls. If a dial peer is dedicated to local calls only, disable the roaming capability.
The roaming dial peer must work with a roaming service provider. If the dial peer allows a roaming user to go through, and the service provider is not roaming-enabled, the call fails.
Examples
The following example enables the roaming capability for the dial peer:
Related Commands
Command
|
Description
|
roaming (settlement)
|
Enables the roaming capability for a settlement provider.
|
settle-call
|
Limits the dial peer to using only the specific clearinghouse identified by the specified provider-number.
|
settlement roam-pattern
|
Configures a pattern to match against when determining roaming.
|
roaming (settlement)
To enable the roaming capability for a settlement provider, use the roaming command in settlement configuration mode. To disable the roaming capability, use the no form of this command.
roaming
no roaming
Syntax Description
This command has no arguments or keywords.
Defaults
No roaming
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Enable roaming capability of a settlement provider if that provider can authenticate a roaming user and route roaming calls.
A roaming call is successful only if both the settlement provider and the outbound dial peer for that call are roaming-enabled.
Examples
The following example enables the roaming capability for the settlement provider:
Related Commands
Command
|
Description
|
roaming (dial-peer mode)
|
Enables the roaming capability for the dial peer.
|
settle-call
|
Limits the dial peer to using only the specific clearinghouse identified by the specified provider-number.
|
settlement roam-pattern
|
Configures a pattern to match against when determining roaming.
|
rtsp client session history duration
To specify how long to keep Real Time Streaming Protocol (RTSP) session history records in memory, use the rtsp client session history duration command in global configuration mode. To set the value to the default, use the no form of this command.
rtsp client session history duration number
no rtsp client session history duration number
Syntax Description
number
|
Specifies how long, in minutes, to keep the record.
|
Defaults
10 minutes
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Examples
The following example sets the RTSP session history to 500 minutes:
rtsp client session history duration 500
Related Commands
Command
|
Description
|
call application voice load
|
Allows reload of an aplication that was loaded via the MGCP scripting package.
|
rtsp client session history records
|
Specifies the number of RTSP client session history records kept during the session.
|
show call application voice
|
Displays all TCL or MGCP scripts that are loaded.
|
show rtsp client session
|
Displays cumulative information about the RTSP session records.
|
rtsp client session history records
To configure the number of records to keep in the RTSP client session history, use the rtsp client session history records command in global configuration mode. To set the value to the default, use the no form of this command.
rtsp client session history records number
no rtsp client session history records number
Syntax Description
number
|
Specifies the number of records to retain in a session history. Values range from 1 to 100000.
|
Defaults
50 records
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Examples
The following example sets the RTSP client history to 500 records:
rtsp client session history records 500
Related Commands
Command
|
Description
|
call application voice load
|
Allows reload of an aplication that was loaded via the MGCP scripting package.
|
rtsp client session history duration
|
Specifies the how long the RTSP is kept during the session.
|
show call application voice
|
Displays all TCL or MGCP scripts that are loaded.
|
rule
To apply a translation rule to a calling party number or a called party number for both incoming and outgoing calls, use the rule command in translation-rule configuration mode. To remove the translation rule, use the no form of this command.
rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]
no rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]
Syntax Description
name-tag
|
The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. Range is from 1 through 2147483647.
|
input-matched-pattern
|
The input string of digits for which pattern matching is performed.
|
substituted-pattern
|
The replacement digit string that results after pattern matching is performed. Regular expressions are used to carry out this process.
|
match-type
|
(Optional) The choices for this field are international, national, subscriber, abbreviated, unknown, and any, as defined by the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Q.931 specification. If you enter the match-type value, then you must also enter the substituted-type value.
|
substituted-type
|
(Optional) The choices for this field are international, national, subscriber, abbreviated and unknown, as defined by the ITU Q.931 specification.
|

Note
In the syntax description above, the square brackets indicate optional values. When using this command, do not include these square brackets as part of the syntax. They are not valid parameters in the rule command. The square brackets can only be used in actual syntax for such commands as the destination-pattern and incoming called-number commands, where the syntax specifically allows this delimiter.
Defaults
No default behavior or values.
Command Modes
Translation-rule configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300 universal access server.
|
12.0(7)XKs
|
This command was first supported for Voice over IP on the following platforms: Cisco 2600 and 3600 series routers and Cisco MC3810 multiservice concentrators.
|
12.1(1)T
|
This command was first supported on the T train for Voice over IP on the following platforms: Cisco 1750 routers, Cisco 2600 and 3600 series routers, Cisco AS5300 universal access servers, Cisco 7200 series routers, and Cisco 7500 series routers.
|
12.1(2)T
|
This command was first supported for Voice over IP on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
When configuring your dial peers, you are provided with an option called the translation rule. This option applies a translation rule to a calling party number (Automatic Number Identification [ANI]) or a called party number (Dial Number Information Service [DNIS]) for both incoming and outgoing calls within Cisco H.323 voice-enabled gateways. Also, the rule allows translation of the type of number.
Examples
The following example applies a translation rule. If a called number starts with 5552205 or 52205, then translation rule 21 will use the rule command to forward the number to 14085552205 instead.
rule 1 555.% 1408555 subscriber international
rule 2 7.% 1408555 abbreviated international
In the next example, if a called number is either 14085552205 or 014085552205, then after the execution of the translation rule 345, the forwarding digits will be 52205. If the match type is configured and the type is not "unknown," then the dial peer matching will be required to match input string numbering type.
rule 1 .%555.% 7 any abbreviated
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
test translation-rule
|
Tests the execution of the translation rules on a specific name tag.
|
translate
|
Applies a translation rule to a calling party number or a called party number for incoming calls
|
translate-outgoing
|
Applies a translation rule to a calling party number or a called party number for outgoing calls
|
translation-rule
|
Creates a translation name and enters translation-rule configuration mode.
|
voip-incoming translation-rule
|
Captures calls that originate from H.323-compatible clients.
|
security
To enable authentication and authorization on a gatekeeper, use the security command in gatekeeper configuration mode. To disable security, use the no form of this command.
security {any | h323-id | e164} {password default password | password separator character}
no security {any | h323-id | e164} {password default password | password separator character}
Syntax Description
any
|
Uses the first alias of an incoming registration, admission, and status (RAS) protocol registration, regardless of its type, as the means of identifying the user to RADIUS/TACACS+.
|
h323-id
|
Uses the first H.323 ID type alias as the means of identifying the user to RADIUS/TACACS+.
|
e164
|
Uses the first E.164 address type alias as the means of identifying the user to RADIUS/TACACS+.
|
password default password
|
Specifies the default password that the gatekeeper associates with endpoints when authenticating them with an authentication server. The password must be identical to the password on the authentication server.
|
password separator character
|
Specifies the character that endpoints use to separate the H.323-ID from the piggybacked password in the registration. Specifying this character allows each endpoint to supply a user-specific password. The separator character and password will be stripped from the string before it is treated as an H.323-ID alias to be registered.
Note that passwords may only be piggybacked in the H.323-ID, not the E.164 address, because the E.164 address allows a limited set of mostly numeric characters. If the endpoint does not wish to register an H.323-ID, it can still supply an H.323-ID consisting of just the separator character and password. This H.323-ID consisting of just the separator character and password will be understood to be a password mechanism and no H.323-ID will be registered.
|
Defaults
No default
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers.
|
Usage Guidelines
Use the security command to enable identification of registered aliases by RADIUS/TACACS+. If the alias does not exist in RADIUS/TACACS+, the endpoint will not be allowed to register.
A RADIUS/TACACS+ server and encryption key must have been configured in Cisco IOS software for security to work.
Only the first alias of the proper type will be identified. If no alias of the proper type is found, the registration will be rejected.
This command does not allow you to define the password mechanism unless the security type (h323-id or e164 or any) has been defined. Although the no security password command undefines the password mechanism, it leaves the security type unchanged, so security is still enabled. However, the no security command disables security entirely, including removing any existing password definitions.
Examples
The following example enables identification of registrations using the first H.323 ID found in any registration:
The following example enables security, authenticating all users by using their H.323-IDs and a password of qwerty2x:
security password qwerty2x
The next example enables security, authenticating all users by using their H.323-IDs and the password entered by the user in the H.323-ID alias he or she registers:
security password separator !
Now if a user registers with an H.323-ID of joe!024aqx, the gatekeeper authenticates user joe with password 024aqx, and if that is successful, registers the user with the H.323-ID of joe. If the exclamation point is not found, the user is authenticated with the default password, or a null password if no default has been configured.
The following example enables security, authenticating all users by using their E.164 IDs and the password entered by the user in the H.323-ID alias he or she registers:
security password separator !
Now if a user registers with an E.164 address of 5551212 and an H.323-ID of !hs8473q6, the gatekeeper authenticates user 5551212 and password hs8473q6. Because the H.323-ID string supplied by the user begins with the separator character, no H.323-ID is registered, and the user is known only by the E.164 address.
Related Commands
Command
|
Description
|
accounting (gatekeeper)
|
Enables the accounting security feature on the gatekeeper.
|
radius-server host
|
Specifies a RADIUS server host.
|
radius-server key
|
Sets the authentication and encryption key for all RADIUS communications between the router and the RADIUS daemon.
|
sequence-numbers
To enable the generation of sequence numbers in each frame generated by the digital signal processor (DSP) for Voice over Frame Relay applications, use the sequence-numbers command in dial-peer configuration mode. To disable the generation of sequence numbers, use the no form of this command.
sequence-numbers
no sequence-numbers
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco 2600 and 3600 series routers and the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated into the Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
Sequence numbers on voice packets allow the digital signal processor (DSP) at the playout side to detect lost packets, duplicate packets, or out-of-sequence packets. This helps the DSP to mask out occasional drop-outs in voice transmission at the cost of one extra byte per packet. The benefit of using sequence numbers versus the cost in bandwidth of adding an extra byte to each voice packet on the Frame Relay network must be weighed to determine whether to disable this function for your application.
Another factor to consider is that this command does not affect codecs that require a sequence number, such as G.726. If you are using a codec that requires a sequence number, the DSP will generate one regardless of the configuration of this command.
Examples
The following example shows how to disable the generation of sequence numbers for VoFR frames on a Cisco 2600 series or 3600 series router or on a Cisco MC3810 multiservice concentrator for VoFR dial peer 200, starting from global configuration mode:
Related Commands
Command
|
Description
|
called-number (dial-peer)
|
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
|
cptone
|
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dtmf-relay (Voice over Frame Relay)
|
Enables the generation of FRF.11 Annex A frames for a dial peer.
|
session protocol (Voice over Frame Relay)
|
Establishes a session protocol for calls between the local and remote routers via the packet network.
|
session target
|
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
server (RLM)
To identify an RLM server, use the server RLM configuration command. To remove the identification, use the no form of this command
server name-tag
no server name-tag
Syntax Description
name-tag
|
Name to identify the server configuration so that multiple entries of server configuration can be entered.
|
Defaults
Disabled
Command Modes
RLM configuration
Command History
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Usage Guidelines
Each server can have multiple entries of IP addresses or aliases.
Examples
The following example identifies the RLM server and defines the associated IP addresses:
rlm group 1
server r1-server
link address 10.1.4.1 source Loopback1 weight 4
link address 10.1.4.2 source Loopback2 weight 3
Related Commands
Command
|
Description
|
clear interface
|
Resets the hardware logic on an interface.
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
interface
|
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
show rlm group statistics
|
Displays the network latency of the RLM group.
|
show rlm group status
|
Displays the status of the RLM group.
|
show rlm group timer
|
Displays the current RLM group timer values.
|
shutdown (RLM)
|
Shuts down all of the links under the RLM group.
|
timer
|
Overwrites the default setting of timeout values.
|
server registration-port
To configure the listener port for the server to establish a connection with the gatekeeper, use the server registration-port command in gatekeeper configuration mode. To force the gatekeeper to close the listening socket so that no more new registration takes place, use the no form of this command.
server registration-port port number
no server registration-port port number
Syntax Description
port number
|
Specifies a single range of values from 1 through 65535 for the port number on which the gatekeeper listens for external server connections.
|
Defaults
The registration port of the gatekeeper is not configured, so no external server can register with this gatekeeper.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Use this command to configure a server registration port to poll for servers that want to establish connections with the gatekeeper on this router.
Note
The no form of this command forces the gatekeeper on this router to close the listen socket, so it cannot accept more registrations. However, existing connections between the gatekeeper and servers are left open.
Examples
The following example shows how a listener port for a server is established for connection with a gatekeeper:
server registration-port 20000
Related Commands
Command
|
Description
|
server trigger
|
Configure static server triggers for specific RAS messages to be forwarded to a specified server.
|
server trigger
To configure a static server trigger for external applications, use the server trigger command in gatekeeper configuration mode. To remove a single statically configured trigger entry, use the no form of this command. To remove every static trigger you configured if you want to delete them all, use the all keyword.
server trigger {arq | lcf | lrj | lrq | rrq | urq} gkid priority server-id server-ipaddress server-port
no server trigger {arq | lcf | lrj | lrq | rrq | urq} gkid priority
no server trigger all
Syntax Description
all
|
Specified to delete all command-line interface configured triggers.
|
arq, lcf, lrj, lrq, rrq, urq
|
Registration, admission, and status (RAS) protocol message types. Use these message types to specify a submode in the gatekeeper configuration mode in which you configure a trigger for the gatekeeper to act upon. Specify only one message type per server trigger command. There is a different trigger submode for each message type. Each trigger submode has its own set of applicable commands.
|
gkid
|
The local gatekeeper identifier.
|
priority
|
The priority for each trigger. The range is from 1 through 20, with 1 being the highest priority.
|
server-id
|
The ID number of the external application.
|
server-ipaddress
|
The IP address of the server.
|
server-port
|
The port on which the Cisco IOS gatekeeper listens for messages from the external server connection.
|
Defaults
No server triggers are set.
Command Modes
Gatekeeper configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Use this command to configure a static server trigger. There are six different server triggers—one for each of the RAS messages. To configure a trigger, go to its submode where a set of subcommands are used to trigger a condition. See the following examples.
In ARQ submode, enter the following syntax:
server trigger arq gkid priority server-id server-ipaddress server-port
In LCF submode, enter the following syntax:
server trigger lcf gkid priority server-id server-ipaddress server-port
In LRJ submode, enter the following syntax:
server trigger lrj gkid priority server-id server-ipaddress server-port
In LRQ submode, enter the following syntax:
server trigger lrq gkid priority server-id server-ipaddress server-port
In RRQ submode, enter the following syntax:
server trigger rrq gkid priority server-id server-ipaddress server-port
In URQ submode, enter the following syntax:
server trigger urq gkid priority server-id server-ipaddress server-port
The following options are available in all submodes:
info-only
|
Information only—no need to wait for acknowledgment.
|
shutdown
|
Enter this subcommand to temporarily disable a trigger. The gatekeeper does not consult triggers in a shutdown state when determining what message to forward.
|
The destination-info argument is under the ARQ, LRQ, LCF, and LRJ submode and has the following options:
destination-info
|
Configure destination-info to trigger one of the following conditions:
|
e164
email-id
h323-id
word
|
Configure an E.164 pattern.
|
Configure an email ID.
|
Configure an H.323 ID.
|
When configuring the e164 address option, the email-id option, or the h323-id option above, the E.164 address can end in a trailing `., `s, or `*'.
|
The redirect-reason argument is under the ARQ and LRQ submodes and has the following options:
redirect-reason
|
Configure a redirect-reason to trigger on (range of 0 through 65535) with the following reserved values:
|
0
1
2
4
9
10
15
|
Unknown reason.
|
Call forwarding busy or called DTE busy.
|
Call forwarded no reply.
|
Call deflection.
|
Called DTE out of order.
|
Call forwarding by the call DTE.
|
Call forwarding unconditionally.
|
The remote-ext-address argument is under the LCF trigger submode and has the following options:
remote-ext-address
|
Configure remote extension addresses, with the following options:
|
e164
word
|
Configure an E.164 pattern.
|
When configuring the e164 address option, the email-id option, or the h323-id option above, the E.164 address can end in a trailing `., `s, or `*'.
|
The endpoint-type argument is under the RRQ and URQ trigger submodes and has the following options:
endpoint-type
|
Configure the type of endpoint to trigger, with the following options:
|
gatekeeper
h320-gateway
mcu
other-gateway
|
The endpoint is an H.323 gatekeeper.
|
The endpoint is an H.320 gateway.
|
The endpoint is a multipoint control unit (MCU).
|
The endpoint is another type of gateway not specified on this list.
|
proxy
terminal
voice-gateway
|
The endpoint is a H.323 proxy.
|
The endpoint is an H.323 proxy.
|
The endpoint is a voice gateway.
|
The supported-prefix keyword is under the RRQ and URQ submodes and has the following options:
supported-prefix
|
Configure the gateway technology prefix to trigger on.
|
word
|
Enter a word within the set of "0123456789#*" when configuring the E.164 pattern for a gateway technology prefix.
|
Entering the no form of the server trigger command removes the trigger definition from the Cisco IOS gatekeeper with all statically configured conditions under that trigger.
Examples
The following example configures a server trigger on gatekeeper sj.xyz.com to notify external server "Server-123" of any call to an E.164 number that starts with 1800 followed by any 7 digits (1800551212, for example):
server trigger arq sj.xyz.com 1 Server-123 1.14.93.130 1751
destination-info e164 1800.......
Related Commands
Command
|
Description
|
server registration port
|
Configure a gatekeeper listening port to listen for external server connections.
|
show gatekeeper servers
|
Show a list of currently registered and statically configured triggers on this gatekeeper router.
|
session
To associate a transport session with a specified session-group, use the session group command in backhaul session manager configuration mode. It is assumed that the server is located on a remote machine. To delete the session, use the no form of this command.
session group group-name remote_ip remote_port local_ip local_port priority
no session group group-name remote_ip remote_port local_ip local_port priority
Syntax Description
group
|
Specifies the session-group name.
|
group-name
|
Session-group name.
|
remote_ip
|
Remote IP address.
|
remote_port
|
Remote port number. Range is 1024 through 9999.
|
local_ip
|
Local IP address.
|
local_port
|
Local port number. Range is 1024 through 9999.
|
priority
|
Priority of the session-group. Range is 0 through 9999 and 0 is the highest priority.
|
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850 platform.
|
Examples
To associate a transport session with the session-group Group5 and specify the parameters described above, see the following example:
Router(config-bsm)# session group group5 161.44.2.72 5555 172.18.72.198 5555 1
session protocol
To specify a session protocol for calls between the local and remote routers using the packet network, use the session protocol command in dial-peer configuration mode. To reset the default value for this command, use the no form of this command.
session protocol {cisco | sipv2 | aal2-trunk | smtp}
no session protocol
Syntax Description
cisco
|
Configure the dial peer to use proprietary Cisco VoIP session protocol.
|
sipv2
|
SIP users should use this option. This option configures the VoIP dial peer to use IETF SIP.
|
aal2-trunk
|
AAL2 nonswitched trunk session protocol.
|
smtp
|
Specifies Simple Mail Transfer Protocol (SMTP) session protocol.
|
Defaults
No default behaviors or values.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax on the Cisco AS5300 universal access server.
|
12.1(1)T
|
The sipv2 option was added.
|
12.1(1)XA
|
Support was added for VoATM dial peers on the Cisco MC3810 multiservice concentrator with the aal2-trunk keyword.
|
12.1(2)T
|
Modifications to this command in Cisco IOS Release 12.1(1)XA were integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
The keyword cisco is applicable only to VoIP on the Cisco 3600 series routers. The keyword aal2-trunk is applicable only to VoATM on the Cisco MC3810 multiservice concentrator.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following is an example of configuring a VoIP dial peer for H.323 or SIP as the session protocol for VoIP call signaling:
The following example selects AAL2 trunking as the session protocol on a Cisco MC3810 multiservice concentrator:
session protocol aal2-trunk
The following example selects Cisco Session Protocol as the session protocol on a Cisco 3600 series router:
The following example selects SMTP as the session protocol:
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.
|
session target (VoIP)
|
Configures a network-specific address for a dial peer.
|
session protocol (Voice over Frame Relay)
To establish a Voice over Frame Relay protocol for calls between the local and remote routers via the packet network, use the session protocol command in dial-peer configuration mode. To reset the default value, use the no form of this command.
session protocol {cisco-switched | frf11-trunk}
no session protocol
Syntax Description
cisco-switched
|
Specifies proprietary Cisco VoFR session protocol. (This is the only valid session protocol for the Cisco 7200 series.)
|
frf11-trunk
|
Specifies FRF.11 session protocol.
|
Defaults
cisco-switched
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced for VoIP.
|
12.0(3)XG
|
This command was modified to support VoFR on the Cisco 2600, 3600, and 7200 series routers and the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
The cisco-switched and frf11-trunk keywords were added for VoFR dial peers.
|
Usage Guidelines
For Cisco-to-Cisco dial peer connections, Cisco recommends that you use the default session protocol because of the advantages it offers over a pure FRF.11 implementation. When connecting to FRF.11-compliant equipment from other vendors, use the FRF.11session protocol.
Note
When using the FRF.11 session protocol on Cisco 2600 series and 3600 series routers, you must also use the called-number command.
Examples
The following example shows how to configure the FRF.11 session protocol on a Cisco 2600 series or 3600 series router for VoFR dial peer 200:
session protocol frf11-trunk
The following example shows how to configure the FRF.11 session protocol on a Cisco MC3810 multiservice concentrator for VoFR dial peer 200:
session protocol frf11-trunk
Related Commands
Command
|
Description
|
called-number (dial-peer)
|
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
|
cptone
|
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dtmf-relay (Voice over Frame Relay)
|
Enables the generation of FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred order of a dial peer within a rotary hunt group.
|
session target
|
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session protocol aal2
To enter the voice-service-session configuration mode and specify AAL2 trunking on a Cisco MC3810 multiservice concentrator, use the session protocol aal2 command in voice-service configuration mode.
session protocol aal2
Syntax Description
This command has no keywords or arguments.
Defaults
There is no default setting for this command.
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
This command applies to VoATM on the MC3810 multiservice concentrator.
In the voice-service-session configuration mode for AAL2, you can configure only AAL2 features, such as call admission control and subcell multiplexing.
Examples
The following example shows how to access the voice-service-session configuration mode, beginning in global configuration mode:
session protocol multicast
To set the session protocol as multicast, use the session protocol multicast command dial-peer configuration mode. To negate this command and return to the Cisco default session protocol, use the no version of this command.
session protocol multicast
no session protocol multicast
Syntax Description
There are no keywords or arguments.
Defaults
When this command is not implemented, the default session protocol is cisco.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(2)XH
|
This command was introduced on Cisco 2600 and Cisco 3600 series routers for the Cisco Hoot and Holler over IP application.
|
12.1(3)T
|
This command was integrated into the Cisco IOS Release 12.1(3)T.
|
Usage Guidelines
Use the session protocol multicast dial-peer configuration command for voice conferencing in a
Hoot and Holler networking implementation. This command allows more than two ports to join the same session simultaneously. It is supported on Cisco 2600 and Cisco 3600 series routers.
Examples
The following example shows the use of the session protocol multicast dial-peer configuration command in context with its accompanying commands:
session protocol multicast
session target ipv4:237.111.0.111:22222
Related Commands
Command
|
Description
|
session target ipv4
|
Assigns the session target for voice-multicasting dial peers.
|
session target (VoATM)
To specify a network-specific address for a specified VoATM dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.
Cisco 3600 Series Routers Voice over ATM Dial Peers
session target interface pvc {name | vpi/vci | vci}
no session target
Cisco MC3810 Multiservice Concentrator Voice over ATM Dial Peers
session target {serial | atm} interface pvc {word | vpi/vci | vci} cid
no session target
Syntax Description
serial
|
Specifies the serial interface for the dial-peer address.
|
atm
|
Specifies the ATM interface. The only valid number is 0.
|
interface
|
Interface type and interface number on the router.
|
pvc
|
The specific ATM permanent virtual circuit (PVC) for this dial peer.
|
word
|
(Optional) A name that identifies the PVC. The argument can identify the PVC if a word identifier was assigned when the PVC was created.
|
name
|
The PVC name.
|
vpi/vci
|
ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC.
On the Cisco 3600, if you have the Multiport T1/E1 ATM network module with IMA installed, the valid range for vpi is from 0 to 5, and the valid range for vci is from 1 to 255.
If you have the OC3 ATM Network Module installed, the valid range for vpi is from 0 to 15, and the valid range for vci is from 1 to 1023.
|
vci
|
ATM network virtual channel identifier (VCI) of this PVC.
|
cid
|
ATM network channel identifier (CID) of this PVC. The valid range is from 8 to 255.
|
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
Support was added for VoATM, VoHDLC, and POTS dial peers on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
Support was added for VoATM dial peers on the Cisco 3600 series routers. Support for VoHDLC on the Cisco MC3810 multiservice concentrator was removed.
|
12.1(2)T
|
Support was added for VoATM on Cisco MC3810 multiservice concentrators.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origin and the loopback type selected.
This command applies to on-ramp store-and-forward fax functions.
You must enter the session protocol aal2-trunk dial-peer configuration command before you can specify a cid for a dial peer for VoATM on the Cisco MC3810 multiservice concentrator.
Note
This command does not apply to plain old telephone service (POTS) dial peers.
Examples
The following example configures a session target for Voice over ATM on a Cisco MC3810 multiservice concentrator. The session target is sent to ATM interface 0, and for a PVC with a VCI of 20.
destination-pattern 13102221111
session target atm0 pvc 20
The following example delivers fax-mail to multiple recipients:
session target marketing-information@mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
The following example displays configuring a session target for Voice over ATM on the Cisco 3600 series. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.
destination-pattern 13102221111
session target atm1/0 pvc 1/100
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
destination-pattern
|
Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
session protocol
|
Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session target (VoFR)
To specify a network-specific address for a specified VoFR dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.
Cisco 2600 and 3600 Series Routers Voice over Frame Relay Dial Peers
session target interface dlci [cid]
no session target
Cisco MC3810 Multiservice Concentrator Voice over Frame Relay Dial Peers
session target interface dlci [cid]
no session target
Cisco 7200 Series Routers Voice over Frame Relay Dial Peers
session target interface dlci
no session target
Syntax Description
interface
|
Specifies the serial interface and interface number (slot number and port number) associated with this dial peer. For the range of valid interface numbers for the selected interface type, enter a ? character after the interface type.
|
dlci
|
Specifies the data link connection identifier for this dial peer. The valid range is from 16 to 1007.
|
cid
|
(Optional) Specifies the DLCI subchannel to be used for data on FRF.11 calls. A CID must be specified only when the session protocol is frf11-trunk. When the session protocol is cisco-switched, the CID is dynamically allocated. The valid range is from 4 to 255.
Note By default, CID 4 is used for data; CID 5 is used for call-control. We recommend that you select CID values between 6 and 63 for voice traffic. If the CID is greater than 63, the FRF.11 header will contain an extra byte of data.
|
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
Support was added for VoFR, VoHDLC, and POTS dial peers on the Cisco MC3810 multiservice concentrator.
|
12.0(3)XG
|
Support was added for VoFR dial peers on the Cisco 2600 series and 3600 series routers. The cid option was added.
|
12.0(4)T
|
Support was added for VoFR and POTS dial peers on the Cisco 7200 series routers and the support added in Cisco IOS Release 12.0(3)XG was integrated into Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origin and the loopback type selected.
For VoFR dial peers, the cid option is not allowed when the cisco-switched option for the session protocol command is used.
Examples
The following example configures a session target for Voice over Frame Relay on a Cisco MC3810 multiservice concentrator with a session target on serial port1 and a DLCI of 200:
destination-pattern 13102221111
session target serial1 200
The following example shows how to configure serial interface 1/0, DLCI 100 as the session target for VoFR dial peer 200 (an FRF.11 dial peer) on a Cisco 2600 series or 3600 series router, starting from global configuration mode and using the FRF.11 session protocol:
destination-pattern 13102221111
session protocol frf11-trunk
session target serial 1/0 100 20
The following example delivers fax-mail to multiple recipients:
session target marketing-information@mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
destination-pattern
|
Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
session protocol
|
Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session target (VoIP)
To specify a network-specific address for a specified VoIP dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.
Cisco 2600 and Cisco 3600 Series Routers and Cisco MC8310 Multiservice Concentrator Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed | ras | settlement}
no session target
Cisco AS5300 Universal Access Server Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed | mailto: | {name | $d$}@domain-name |
ipv4:destination-address | dns:[$s$. | $d$. | $u$. | $e$.] host-name}
no session target
Cisco AS5800 Universal Access Server Voice over IP Dial Peers
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp
| loopback:compressed | loopback:uncompressed}
no session target
Syntax Description
ipv4:destination-address
|
IP address of the dial peer.
|
dns:[$s$...] host-name
|
Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device.
(Optional) Use one of the following three wildcards with this keyword when defining the session target for Voice over IP (VoIP) peers:
$s$.—Indicates that the source destination pattern will be used as part of the domain name.
$d$.—Indicates that the destination number will be used as part of the domain name.
$e$.—Indicates that the digits in the called number will be reversed, periods will be added between the digits of the called number, and this string will be used as part of the domain name.
$u$.—Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name.
|
loopback:rtp
|
Indicates that all voice data will be looped back to the source. This is applicable for VoIP peers.
|
loopback:compressed
|
Indicates that all voice data will be looped back in compressed mode to the source. This is applicable for POTS peers.
|
loopback:uncompressed
|
Indicates that all voice data will be looped-back in uncompressed mode to the source. This is applicable for POTS peers.
|
ras
|
Indicates that the registration, admission, and status (RAS) signaling function protocol is being used, meaning that a gatekeeper will be consulted to translate the E.164 address into an IP address.
|
settlement provider-number
|
Indicates that the settlement server is the target to resolve the terminating gateway address. Enter the provider IP address for provider number.
|
Defaults
The default state for this command is enabled, with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers.
|
12.0(3)T
|
Support was added for VoIP and POTS dial peers on the Cisco AS5300 universal access server. The parameter was added for RAS.
|
12.0(4)XJ
|
Support was added for store-and-forward fax on the Cisco AS5300 universal access server platform.
|
12.1(1)T
|
Support was added for session target type of settlement.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origin and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you must configure if you have groups of numbers associated with a particular router.
Use the session target ras command to specify that the RAS protocol is being used to determine the IP address of the session target.
In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command. When configuring the VoIP dial peers for a settlement server, if session target type is settlement, the provider-number parameter in the session target and settle-call commands should be identical.
When the VoIP dial peers are configured for a settlement server, if the session target type is settlement, the provider-number parameter in the session target and settle-call commands should be identical.
Examples
The following example configures a session target using DNS for a host, "voice_router," in the domain cisco.com:
session target dns:voice_router.cisco.com
The following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify the dial peer. As in the preceding example, the domain is "cisco.com."
destination-pattern 1310222....
session target dns:$u$.cisco.com
The following example configures a session target using DNS, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
destination-pattern 13102221111
session target dns:$d$.cisco.com
The following example configures a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
destination-pattern 12345
session target dns:$e$.cisco.com
The following example configures a session target using RAS:
destination-pattern 13102221111
The following example configures a session target using settlement:
session target settlement:0
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial-peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
destination-pattern
|
Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
|
session protocol
|
Establishes a session protocol for calls between the local and remote routers through the packet network in Voice over IP.
|
settle-call
|
Specifies that settlement is to be used for this dial peer, regardless of session target type.
|
session transport
To configure the VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in dial-peer configuration mode. To reset the value to the default, use the no form of this command.
session transport {udp | tcp }
Syntax Description
udp
|
Configure the SIP dial peer to use the UDP transport layer protocol. This is the default.
|
tcp
|
Configure the SIP dial peer to use the TCP transport layer protocol.
|
Defaults
The SIP dial peer uses UDP.
Note
The transport protocol specified with the transport command and the one specified with the session transport command must be the same.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use show sip-ua status to ensure that the transport protocol that you set using the session transport command matches the protocol set using the transport command.
Examples
The following example shows a VoIP dial peer configured to use UDP as the underlying transport
layer protocol for SIP messages:
set
To create a fault-tolerant or non-fault-tolerant session-set with the client or server option, use the set command in backhaul session manager configuration mode. To delete the set, use the no form of this command.
set set-name { client | server } { ft | nft }
no set set-name { client | server } { ft | nft }
Syntax Description
set-name
|
Session-set name.
|
client
|
Client option. The session-set should only be configured as client for backhaul.
|
server
|
Server option.
|
ft
|
Fault-tolerant. Fault-tolerance is the level of ability within a system to operate properly even if a group in the set fails.
|
nft
|
Non-fault-tolerant. Only one group is allowed in a non-fault-tolerant set.
|
Defaults
No default behavior or values.
Command Modes
Backhaul session manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Usage Guidelines
There can be multiple groups associated with a session-set.
The session-set should only be configured for the client for backhaul (not the server).
A set cannot be deleted unless the groups associated with the set are deleted first.
Examples
To specify the client set named Set1 to fault-tolerant, see the following example:
Router(config-bsm)# set set1 client ft
settle-call
To force a call to be authorized with a settlement server that uses the address resolution method specified in the session target type command, use the settle-call command in dial-peer configuration mode. To make sure that no authorization will be performed by a settlement server, use the no form of this command.
settle-call provider-number
no settle-call provider-number
Syntax Description
provider-number
|
Digit defining the ID of a particular settlement server. The only valid entry is 0.
Note If session target type is settlement, the provider-number argument in the session target and settle-call commands should be identical.
|
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Using the session target command, a dial peer can determine the address of the terminating gateway through the ipv4, dns, ras, and settlement keywords.
If the session target is not settlement, and the settle-call provider-number argument is set, the gateway resolves the terminating gateway's address using the specified method and then requests the settlement server to authorize that address and create a settlement token for that particular address. If the server cannot authorize the terminating gateway address suggested by the gateway, the call fails.
Do not combine the session target types ras and settle-call. Combination of session target types is not supported in Cisco IOS Release 12.1(1)T.
Examples
The following example sets a call to be authorized with a settlement server that uses the address resolution method specified in the session target:
destination-pattern 1408.......
session target ipv4:172.22.95.14
Related Commands
Command
|
Description
|
session target
|
Specifies a network-specific address for a specified dial peer.
|
settlement
To enter settlement configuration mode and specify the attributes specific to a settlement provider, use the settlement command in global configuration mode. To disable the settlement provider, use the no form of this command.
settlement provider-number
no settlement provider-number
Syntax Description
provider-number
|
Specifies a digit that defines a particular settlement server. The only valid entry is 0.
|
Defaults
0
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The variable provider-number defines a particular settlement provider. For Cisco IOS Release 12.1, only one clearinghouse per system is allowed, and the only valid value for provider-number is 0.
Examples
This example shows how to enter settlement configuration mode:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the length of time for which a connection is maintained after a communication exchange is completed.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
retry-limit
|
Sets the connection retry limit.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown
|
Brings up the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
settlement roam-pattern
To configure a pattern that must be matched to determine if a user is roaming, use the settlement roam-pattern command in global configuration mode. To delete a particular pattern, use the no form of this command.
settlement provider-number roam-pattern pattern {roaming | no roaming}
no settlement provider-number roam-pattern pattern {roaming | no roaming}
Syntax Description
provider-number
|
Digit defining the ID of particular settlement server. The only valid entry is 0.
|
pattern
|
Specifies a user account pattern.
|
roaming | no roaming
|
Determines whether a user is roaming.
|
Defaults
No default pattern
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
Usage Guidelines
Multiple "roam patterns" could be entered on one gateway.
Examples
The following example will configure a pattern that determines if a user is roaming:
settlement 0 roam-pattern 1222 roam
settlement 0 roam-pattern 1333 noroam
settlement roam-pattern 1444 roam
settlement roam-pattern 1555 noroam
Related Commands
Command
|
Description
|
roaming (settlement)
|
Enables the roaming capability for a settlement provider.
|
settlement
|
Enters settlement configuration mode.
|
sgcp
To start and allocate resources for the Simple Gateway Control Protocol (SGCP) daemon, use the sgcp command in global configuration mode. To terminate all calls, release all allocated resources, and kill the SGCP daemon, use the no form of this command.
sgcp
no sgcp
Syntax Description
This command has no arguments or keywords.
Defaults
The SGCP daemon is not enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
When the SGCP daemon is not active, all SGCP messages are ignored.
When you enter the no sgcp command, the SGCP process is removed.
Note
After you enter the no sgcp command, you must save the configuration and reboot the router for the disabling of SGCP to take effect.
Examples
The following example shows the SGCP daemon being enabled:
The following example shows the SGCP daemon being disabled:
Related Commands
Command
|
Description
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp call-agent
To define the IP address of the default Simple Gateway Control Protocol (SGCP) call agent in the router configuration file, use the sgcp call-agent command in global configuration mode. To remove the IP address of the default SGCP call agent from the router configuration, use the no form of this command.
sgcp call-agent ipaddress [:udp port]
no sgcp call-agent ipaddress
Syntax Description
ipaddress
|
Specifies the IP address or hostname of the call agent.
|
:udp port
|
(Optional) Specifies the UDP port of the call agent.
|
Defaults
No IP address is configured.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Setting this command defines the IP address of the default SGCP call agent to which the router sends an initial RSIP (Restart In Progress) packet when the router boots up. This is used for initial boot-up only before the SGCP call agent contacts the router acting as the gateway.
When you enter the no sgcp call-agent command, only the IP address of the default SGCP call agent is removed.
Examples
The following example shows SGCP being enabled and the IP address of the call agent being specified:
sgcp call-agent 209.165.200.225
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp graceful-shutdown
To block all new calls and gracefully terminate all existing calls (wait for the caller to end the call), use the sgcp graceful-shutdown command in global configuration mode. To unblock all calls and allow new calls to go through, use the no form of this command.
sgcp graceful-shutdown
no sgcp graceful-shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Once you issue this command, all requests for new connections (CreateConnection requests) are denied. All existing calls are maintained until users terminate them, or until you enter the no sgcp command. When the last active call is terminated, the SGCP daemon is terminated, and all resources allocated to it are released.
Examples
The following example shows all new calls being blocked and existing calls being terminated:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp max-waiting-delay
To set the Simple Gateway Control Protocol (SGCP) maximum waiting delay to prevent restart avalanches, use the sgcp max-waiting-delay command in global configuration mode. To restore the default value, use the no form of this command.
sgcp max-waiting-delay delay
no sgcp max-waiting-delay delay
Syntax Description
delay
|
Sets the maximum waiting delay (MWD) value in milliseconds. The valid range is from 0 to 600,000. The default is 3000.
|
Defaults
3,000 milliseconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only, and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Examples
The following example shows the maximum wait delay value set to 40 milliseconds:
sgcp max-waiting-delay 40
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp modem passthru
To enable Simple Gateway Control Protocol (SGCP) modem or fax pass-through, use the sgcp modem passthru command in global configuration mode. To disable SGCP modem or fax pass-through, use the no form of this command.
sgcp modem passthru {ca | cisco | nse}
no sgcp modem passthru {ca | cisco | nse}
Syntax Description
ca
|
Uses the call agent controlled modem upspeed method violation message.
|
cisco
|
Uses a Cisco-proprietary upspeed method based on the protocol.
|
nse
|
Uses the NSE-based modem upspeed method.
|
Defaults
SGCP modem or fax pass-through is disabled by default.
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and Cisco 3600 series routers (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
You can use this command for fax pass-through because the answer tone can come from either modem or fax transmissions. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions.
If you use the nse option, you must also configure the sgcp tse payload command.
Examples
The following example shows SGCP modem pass-through configured using the call agent upspeed method:
The following example shows SGCP modem pass-through configured using the proprietary Cisco upspeed method:
sgcp modem passthru cisco
The following example shows SGCP modem pass-through configured using the NSE-based modem upspeed:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp quarantine-buffer disable
To disable the Simple Gateway Control Protocol (SGCP) quarantine buffer, use the sgcp quarantine-buffer disable command in global configuration mode. To reenable the SGCP quarantine buffer, use the no form of this command.
sgcp quarantine-buffer disable
no sgcp quarantine-buffer disable
Syntax Description
This command has no arguments or keywords.
Defaults
The SGCP quarantine buffer is enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
The SGCP quarantine buffer is the mechanism for buffering the SGCP events between two RQNT messages.
Examples
The following example shows the SGCP quarantine buffer being disabled:
sgcp quarantine-buffer disable
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp request retries
To specify the number of times to retry sending "notify" and "delete" messages to the Simple Gateway Control Protocol (SGCP) call agent, use the sgcp request retries command in global configuration mode. To restore the default value, use the no form of this command.
sgcp request retries count
no sgcp request retries
Syntax Description
count
|
Specifies the number of times a "notify" and "delete" message is retransmitted to the SGCP call agent before it is dropped. The valid range is from 1 to 100. The default is 3.
|
Defaults
The default for the number of times a "notify" and "delete" message is retransmitted to the SGCP call agent before it is dropped is 3
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
The actual retry count may be different from the value you enter for this command. The retry count is also limited by the call agent. If there is no response from the call agent after 30 seconds, the gateway will not retry anymore, even though the number set using the sgcp request retries command has not been reached.
The router will stop sending retries after 30 seconds, regardless of the setting for this command.
Examples
The following example shows the system configured to send the sgcp command 10 times before dropping the request:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp request timeout
To specify how long the system should wait for a response to a request, use the sgcp request timeout command in global configuration mode. To restore the default value, use the no form of this command.
sgcp request timeout timeout
no sgcp request timeout
Syntax Description
timeout
|
Specifies the number of milliseconds to wait for a response to a request. Valid range is from 1 to 10,000.
|
Defaults
500 milliseconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
This command is used for "notify" and "delete" messages, which are sent to the SGCP call agent.
Examples
The following example shows the system configured to wait 40 milliseconds for a reply to a request:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp restart
To trigger the router to send a Restart in Progress (RSIP) message to the Simple Gateway Control Protocol (SGCP) call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller, use the sgcp restart command in global configuration mode. To restore the default value, use the no form of this command.
sgcp restart {delay delay | notify}
no sgcp restart {delay delay | notify}
Syntax Description
delay delay
|
Specifies the restart delay timer value in milliseconds. The valid range is from 0 to 600, and the default value is 0.
|
notify
|
Enables the restart notification upon the SGCP/digital interface state transition.
|
Defaults
Zero (0)
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and Cisco 3600 series routers (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
This command is used to send RSIP messages from the router to the SGCP call agent. The RSIP messages are used to synchronize the router and the call agent. RSIP messages are also sent when the sgcp command is entered to enable the SGCP daemon.
You must enter the notify option to enable RSIP messages to be sent.
Examples
The following example shows the system configured to wait 40 milliseconds before restarting SGCP:
The following example shows the system configured to send an RSIP notification to the SGCP call agent when the T1 controller state changes:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp retransmit timer
To configure the Simple Gateway Control Protocol (SGCP) retransmission timer to use a random algorithm, use the sgcp retransmit timer command in global configuration mode. To restore the default value, use the no form of this command.
sgcp retransmit timer {random}
no sgcp retransmit timer {random}
Syntax Description
random
|
Enables the SGCP retransmission timer to use a random algorithm.
|
Defaults
The SGCP retransmission timer does not use the random algorithm.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 3600 and Cisco MC3810 multiservice concentrator in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Use this command to enable the random algorithm component of the retransmission timer. For example, if the retransmission timer is set to 200 milliseconds, the first retransmission timer is 200 milliseconds, but the second retransmission timer picks up a timer value randomly between either 200 or 400. The third retransmission timer picks up a timer value randomly of 200, 400, or 800 as shown below:
•
First retransmission timer: 200
•
Second retransmission timer: 200 or 400
•
Third retransmission timer: 200, 400, or 800
•
Fourth retransmission timer: 200, 400, 800, or 1600
•
Fifth retransmission timer: 200, 400, 800, 1600, or 3200 and so on.
After 30 seconds, the retransmission timer no longer retries.
Examples
The following example shows the retransmission timer set to use the random algorithm:
sgcp retransmit timer random
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband Tse for fax/modem operation.
|
sgcp timer
To configure how the gateway detects the Real-Time Transport Protocol (RTP) stream lost, use the sgcp timer command in global configuration mode. To restore the default value, use the no form of this command.
sgcp timer {receive-rtcp timer | rtp-nse timer}
no sgcp timer {receive-rtcp timer | rtp-nse timer}
Syntax Description
receive-rtcp timer
|
Sets the multiples of the RTP Control Protocol (RTCP) transmission interval in milliseconds. The valid range is from 1 to 100, and the default is 5.
|
rtp-nse timer
|
Sets the multiples of the RTP named signaling event (NSE) timeout in milliseconds. The valid range is from 100 to 3000, and the default is 200.
|
Defaults
Default for receive-rtcp timer is 5.
Default for rtp-nse timer is 200.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
The RTP NSE timer is used for proxy ringing (the ringback tone is provided at the originating gateway).
Examples
The following example shows the receive-rtcp timer set to 100 milliseconds:
sgcp timer receive-rtcp 100
The following example shows the rtp-nse timer set to 1000 milliseconds:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp tse payload
To enable Inband Telephony Signaling Events (TSE) for fax and modem operation, use the sgcp tse payload command in global configuration mode. To restore the default value, use the no form of this command.
sgcp tse payload type
no sgcp tse payload type
Syntax Description
type
|
Sets the TSE payload type. The valid range is from 96 to 119. The default is 0, meaning that the command is disabled.
|
Defaults
Zero (0)
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 multiservice concentrator and Cisco 3600 series routers (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was integrated into 12.1(2)T and was generally available on the Cisco 3600 series router and the Cisco MC3810 multiservice concentrator
|
Usage Guidelines
Because this command is disabled by default, you must specify a TSE payload type.
If you configure the sgcp modem passthru command to the nse value, then you must configure this command.
Examples
The following example shows the Simple Gateway Control Protocol (SGCP) modem pass-through set using the NSE-based modem upspeed and the Inband Telephony Signaling Events payload value set to 110:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.up or down so that the call agent can synchronize
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
show aal2 profile
To display the ATM adaptation layer 2 (AAL2) profiles configured on the system, use the show aal2 profile command in privileged EXEC mode.
show aal2 profile all | {itut profile-number | custom profile-number | atmf profile-number}
Syntax Description
all
|
Displays International Telecommunication Union Telecommunication Standardization Sector (ITU-T), ATM Forum, and custom AAL2 profiles configured on the system.
|
itut
|
Displays ITU-T profiles configured on the system.
|
profile-number
|
Specifies the profile number of the AAL2 profile to display. The available choices are as follows:
For ITU-T:
• 1 = G.711 u-law
• 2 = G.711 u-law with silence insertion descriptor (SID)
• 7 = G.711 u-law and G.729ar8
For ATMF: None. ATMF is not supported.
For custom:
• 100 = G.711 u-law and G.726r32
• 110 = G.711 u-law, G.726r32, and G.729ar8
|
custom
|
Displays custom profiles configured on the system.
|
atmf
|
Displays ATM Forum profiles configured on the system.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
This command applies to AAL2 Voice over ATM (VoATM) applications on the Cisco MC3810 multiservice concentrator.
Use the show aal2 profile EXEC command to display the AAL2 profiles configured in the system.
Examples
The following is sample output from the show aal2 profile command for displaying all the profiles configured in the system:
Router# show aal2 profile all
Printing all the Profiles in the system
Profile Type: ITUT Profile Number: 1 SID Support: 0
Red enable: 1 Num entries: 1
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Profile Type: ITUT Profile Number: 2 SID Support: 1
Red enable: 1 Num entries: 1
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Profile Type: custom Profile Number: 100 SID Support: 1
Red enable: 1 Num entries: 2
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7
Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15
Profile Type: ITUT Profile Number: 7 SID Support: 1
Red enable: 1 Num entries: 2
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 15
Coding type: g729ar8 Packet length: 10 UUI min: 0 UUI max: 15
Profile Type: custom Profile Number: 110 SID Support: 1
Red enable: 1 Num entries: 3
Coding type: g711ulaw Packet length: 40 UUI min: 0 UUI max: 7
Coding type: g726r32 Packet length: 40 UUI min: 8 UUI max: 15
Coding type: g729ar8 Packet length: 30 UUI min: 8 UUI max: 15
Table 26 provides an alphabetical listing of the fields in this output and a description of each field.
Table 26 show aal2 profile Field Descriptions
Field
|
Description
|
Profile Type
|
Category of codec types configured on DSP. Possible types are ITU-T, ATMF, and custom.
|
ITUT Profile Number
|
Predefined combination of one or more codec types configured for a digital signal processor (DSP).
|
SID Support
|
Silence insertion descriptor.
|
Red enable
|
Redundancy enable for type3 packets.
|
Num entries
|
Number of profile elements.
|
Coding type
|
Voice compression algorithm.
|
Packet length
|
Sample size.
|
UUI min
|
Minimum sequence number on the voice packets.
|
UUI max
|
Maximum sequence number on the voice packets.
|
Related Commands
Command
|
Description
|
codec aal2-profile
|
Sets the codec profile for a DSP on a per-call basis.
|
show atm video-voice address
To display the network service access point (NSAP) address for the ATM interface, enter the show atm video-voice address command in privileged EXEC mode.
show atm video-voice address
Syntax Description
This command has no keywords or arguments.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
Enter this command to review ATM interface NSAP addresses that have been assigned with the atm video aesa command and to ensure that ATM management is confirmed for those addresses.
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays ATM interface NSAP addresses:
Router# show atm video-voice address
nsap address type ilmi status
47.0091810000000002F26D4901.00107B4832E1.FE VOICE_AAL5 Confirmed
47.0091810000000002F26D4901.00107B4832E1.C8 VIDEO_AAL1 Confirmed
Related Commands
Command
|
Description
|
codec aal2-profile
|
Sets the codec profile for a DSP on a per-call basis.
|
show backhaul-session-manager group
To display status, statistics, or configuration information for all available session-groups, use the show backhaul-session-manager group command in privileged EXEC mode.
show backhaul-session-manager group { status | stats | cfg } { all | name group-name }
Syntax Description
status
|
Displays status information for session-groups.
|
stats
|
Displays statistics for session-groups.
|
cfg
|
Displays configuration information for session-groups.
|
all
|
Displays information for all available session-groups.
|
name group-name
|
Displays information for a specific session-group. The group-name argument specifies the name of the session-group.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Examples
The following displays statistics for all session-groups:
Router# show backhaul-session-manager group stats all
Session-Group grp1 statistics
Un-Successful Fail-Over attempts:0
Active Pkts receive count :0
Standby Pkts receive count :0
Total PDUs dispatch err :0
The following displays the current configuration for all session-groups:
Router# show backhaul-session-manager group cfg all
Dest:10.5.0.3 8304 Local:10.1.2.15 8304 Priority:0
Dest:10.5.0.3 8300 Local:10.1.2.15 8300 Priority:2
Dest:10.5.0.3 8303 Local:10.1.2.15 8303 Priority:2
timer cumulative ack :100
timer transfer state :2000
The following displays the current status of all session-groups. This group named grp1 belongs to the set named set1.
The Status will be either Group-OutOfService (no session in the group has been established) or Group-Inservice (at least one session in the group has been established).
The Status(use) will be either Group-Standby (the VSC connected to the other end of this group will go into standby mode), Group-Active (the VSC connected to the other end of this group will be the active VSC), or Group-None (the VSC has not declared its intent yet).
Router# show backhaul-session-manager group status all
Status :Group-OutOfService
Related Commands
Command
|
Description
|
show backhaul-session-manager session
|
Displays status, statistics, or configuration of sessions.
|
show backhaul-session-manager set
|
Displays session-groups associated with a specific or all session-sets.
|
show backhaul-session-manager session
To display various information for about a session or sessions, use the show backhaul-session-manager session command in privileged EXEC mode.
show backhaul-session-manager session { all | ip ip_address }
Syntax Description
all
|
All available sessions.
|
ip ip_address
|
The IP address of the local or remote session.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Examples
To display information for all available sessions, see the following example.
The State will be OPEN (the connection is established), OPEN_WAIT (the connection is awaiting establishment), OPEN_XFER (session failover is in progress for this session, which is a transient state), or CLOSE (this session is down, also a transient state). The session will move to OPEN_WAIT after waiting a fixed amount of time.
The Use-status field indicates whether PRI signaling traffic is currently being transported over this session . The field will be either OOS (this session is not being used to transport signaling traffic) or IS (this session is being used currently to transport all PRI signaling traffic). OOS does not indicate if the connection is established and IS indicates that the connection is established.
Router# show backhaul-session-manager session all
Group:grp1 /*this session belongs to the group named 'grp1' */
Local:10.1.2.15 , port:8303
Remote:10.5.0.3 , port:8303
RUDP Option:Client, Conn Id:0x2
Status:OPEN_WAIT, Use-status:OOS, /*see explanation below */
# of unexpected RUDP transitions (total) 0
# of unexpected RUDP transitions (since last reset) 0
Receive pkts - Total:0 , Since Last Reset:0
Recieve failures - Total:0 ,Since Last Reset:0
Transmit pkts - Total:0, Since Last Reset:0
Transmit Failures (PDU Only)
Due to Blocking (Not an Error) - Total:0, Since Last Reset:0
Due to causes other than Blocking - Total:0, Since Last
Transmit Failures (NON-PDU Only)
Due to Blocking(Not an Error) - Total:0, Since Last Reset:0
Due to causes other than Blocking - Total:0, Since Last
Send window full failures:0
Resource unavailble failures:0
Related Commands
Command
|
Description
|
show backhaul-session-manager group
|
Displays status, statistics, or configuration of a specified or all session-groups.
|
show backhaul-session-manager set
|
Displays session-groups associated with a specified or all session-sets.
|
show backhaul-session-manager set
To display session-groups associated with a specified session-set or all session-sets, use the show backhaul-session-manager set command in privileged EXEC mode.
show backhaul-session-manager set { all | name session-set-name }
Syntax Description
all
|
All available session-sets.
|
name session-set-name
|
A specified session-set.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Examples
To show session groups associated with all session-sets, see the following example:
Router# show backhaul-session-manager set all
Related Commands
Command
|
Description
|
show backhaul-session-manager group
|
Displays status, statistics, or configuration of a specified or all session-groups.
|
show backhaul-session-manager session
|
Displays status, statistics, or configuration of a session or all sessions.
|
show call active
To display active call information for voice calls or fax transmissions in progress, use the show call active command in user EXEC or privileged EXEC mode.
show call active {voice | fax}[brief]
Syntax Description
voice
|
Specifies that information be displayed for all active voice calls.
|
fax
|
Specifies that information be displayed for all active fax calls.
|
brief
|
(Optional) Displays a truncated version of the active call information.
|
Defaults
No default behavior or values.
Command Modes
User EXEC or
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 2600 series and 3600 series.
|
12.0(3)XG
|
Support for VoFR was added.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax on the Cisco AS5300 universal access server.
|
12.0(4)T
|
This command was first supported on the Cisco 7200 series.
|
12.0(7)XK
|
This command was first supported on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.1(3)T
|
This command was modified for Modem Passthrough over VoIP on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use the show call active command to display the contents of the active call table. This command displays information about call times, dial peers, connections, quality of service, and other status and statistical information. If you use the voice keyword, information is displayed about all voice calls currently connected through the router or access server. If you use the fax keyword, information is displayed about all fax calls currently connected.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
See Table 19 for a listing of the information types associated with this command.
Examples
The following is sample output from the show call active voice command:
Router# show call active voice
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
VoiceTxDuration=155310 ms
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
tx_DtmfRelay=inband-voice
SessionTarget=ipv4:1.14.82.14
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
Modem passthrough signaling method is nse:
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 157sec
Time between Buffer Drain/Fills = Min 0sec Max 0sec
The following is sample output from the show call active voice brief command:
Router# show call active voice brief
<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
3 : 104443hs.1 +521 pid:100 Answer 50110 active
dur 00:03:28 tx:20151/3036404 rx:20102/3517936
Tele 0:D:1: tx:199630/199630/0ms g711ulaw noise:-75 acom:11 i/0:-22/-13 dBm
3 : 104648hs.1 +316 pid:2 Originate 55240 active
dur 00:03:28 tx:20102/3276712 rx:20151/3277628
IP 1.14.82.14:18202 rtt:3ms pl:40/0ms lost:0/0/0 delay:67/67/67ms g729r8
MODEMPASS nse buf:0/0 loss 0% 0/0 last 195s dur:0/0s
The following is sample output from the show call active fax command:
Router# show call active fax
ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
The following is sample output from the show call active fax brief command:
Router# show call active fax brief
<ID>: <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state> \
tx:<packets>/<bytes> rx:<packets>/<bytes> <state>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Tele <int>: tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
1 : 22021hs.1 +2263 pid:0 Answer wook song active
IP 0.0.0.0 AcceptedMime:2 DiscardedMime:1
1 : 23193hs.1 +1091 pid:3469 Originate 527.... active
Tele : tx:31200/10910/20290ms noise:-1 acom:-1 i/0:0/0 dBm
Table 27 provides an alphabetical listing of the fields displayed in the output from the show call active command and a description of each field.
Table 27 show call active 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.
|
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.
|
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 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.
|
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.
|
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 or 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.
|
HiWaterPlayoutDelay
|
High-water mark Voice Playout FIFO Delay during this call.
|
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
|
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.
|
Modem passthrough signaling method in use
|
Indicates that this is a modem pass-through call and that named signaling events (NSEs)—also called 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.
|
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.
|
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.
|
RemoteIPAddress
|
Remote system IP address for the VoIP call.
|
RemoteUDPPort
|
Remote system 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 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 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).
|
Time between Buffer Drain/Fills
|
Minimum and maximum durations between jitter buffer drain or fill events, in seconds.
|
TransmitBytes
|
Number of bytes sent by this peer during this call.
|
TransmitPackets
|
Number of packets sent by this peer during this call.
|
TxDuration
|
Duration of transmit path open from this peer to the voice gateway for this call.
|
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.
|
Related Commands
Command
|
Description
|
show call history
|
Displays the call history table.
|
show dial-peer voice
|
Displays configuration information for dial peers.
|
show num-exp
|
Displays how the number expansions are configured in Voice over IP.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show call application voice
To define the names of the audio files that the interactive voice response (IVR) script will play, the operation of the abort keys, the prompts that are used, and caller interaction, use the show call application voice command in EXEC mode.
show call application voice [name | summary]
Syntax Description
name
|
(Optional) The name of the desired IVR application.
|
summary
|
(Optional) Displays a one-line summary. If the command is entered without the summary keyword, a complete detailed description is displayed of the application.
|
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
11.3(6)NA2
|
This command was introduced on the Cisco 2500 series and Cisco 3600 series routers and the Cisco AS5300 universal access server.
|
Usage Guidelines
If the name of a specific application is entered, it will give information about that application.
If the summary keyword is entered, a one-line summary will be displayed about each application.
If the command is entered without the summary, a detailed description of the entered IVR application is displayed.
Examples
This example shows the output for the clid_authen_collect IVR script:
Router# show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
State get_account has 4 actions and 7 events
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
URL: flash:enter_destination.au
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
URL: flash:auth_failed.au
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
Router# show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
State get_account has 4 actions and 7 events
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
URL: flash:enter_destination.au
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
URL: flash:auth_failed.au
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
Related Commands
Command
|
Description
|
call application voice
|
Defines the name to be used for an application and indicates the location of the appropriate IVR script to be used with this application.
|
call application voice load
|
Reloads the designated TCL script.
|
show call fallback cache
To see the current Calculated Planning Impairment Factor (ICPIF) estimates for all IP addresses in cache, use the show call fallback cache command in EXEC mode.
show call fallback cache [ip-address]
Syntax Description
ip-address
|
(Optional) Specifies a specific IP address.
|
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
To clear all entries in the cache, use the clear call fallback cache command.
Examples
The following example displays output from the show call fallback cache command:
Router# show call fallback cache
Probe IP Address Codec Delay Loss ICPIF Reject Accept
----- ---------- ----- ----- ---- ----- ------ ------
1 1.1.1.4 g729r8 40 0 0 0 9
2 122.24.56.25 g729r8 148 10 5 1 4
IP Address IP Address to which the probe is sent
Codec Codec Type of the probe
Delay Delay in milliseconds that the probe incurred
Loss Loss in % that the probe incurred
ICPIF Computed ICPIF value for the probe
Reject Number of times that calls of Codec Type <Codec>
were rejected to the IP Address
Accept Number of times that calls of Codec Type <Codec>
were accepted to the IP Address
active probes Number of destinations being probed
Router# show call fallback cache 10.14.115.53
Probe IP Address Codec ICPIF Reject Accept
----- ---------- ----- ----- ------ ------
1 10.14.115.53 g729r8 0 0 2
Related Commands
Command
|
Description
|
show call fallback stats
|
Displays the call fallback statistics.
|
show call fallback config
To display the call fallback configuration, use the show call fallback config command in EXEC mode.
show call fallback config
Syntax Description
This command has no arguments or keywords.
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Examples
The following example displays output from the show call fallback config command:
Router# show call fallback config
ICPIF value timeout:20 seconds
Number of packets in a probe:20
IP precedence of probe packets:2
Fallback cache size:2 entries
Fallback cache timeout:240 seconds
Instantaneous value weight:65
Related Commands
Command
|
Description
|
call fallback monitor
|
Enables the monitoring of destinations without fallback to alternate dial peers.
|
show voice trunk-conditioning signaling
|
Enables fallback to alternate dial peers in case of network congestion.
|
show call fallback stats
To display the call fallback statistics, use the show call fallback stats command in EXEC mode.
show call fallback stats
Syntax Description
This command has no arguments or keywords.
Defaults
This command is not configured by default.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
To remove all values, use the clear call fallback stats command.
Examples
The following example displays output from the show call fallback stats command:
Router# show call fallback stats
Total accepted calls Number of times that calls were successful over IP.
Total rejected calls Number of times that calls were rejected over IP.
Total cache overflows Number of times that the fallback cache overflowed and requied
pruning.
Related Commands
Command
|
Description
|
clear call fallback stats
|
Clears the call fallback statistics.
|
show call fallback cache
|
Displays the current ICPIF estimates for all IP addresses in the cache.
|
show call history
To display the call history table for voice calls or fax transmissions, use the show call history command in user EXEC or privileged EXEC mode.
show call history {voice | fax}[last number | brief]
Syntax Description
voice
|
Specifies that call history information be displayed for voice calls.
|
fax
|
Specifies that call history information be displayed for fax calls.
|
last number
|
(Optional) Displays the last calls connected, where the number of calls that appear is defined by the number argument. Valid values are from 1 to 100.
|
brief
|
(Optional) Displays a truncated version of the call history table.
|
Defaults
No default behavior or values.
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 for Voice over Frame Relay (VoFR) was added on the Cisco 2600 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 first supported on the Cisco 7200 series.
|
12.0(7)XK
|
Support for the brief keyword was added on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS 12.1(2)T.
|
Usage Guidelines
The show call history command displays a call history table containing a list of voice or 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 between 0 and 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.
When using the fax keyword, this command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following is sample output from the show call history voice command:
Router# show call history voice
DisconnectText=normal call clearing.
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
tx_DtmfRelay=inband-voice
SessionTarget=ipv4:1.14.82.14
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
Modem passthrough signaling method is nse
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
DisconnectText=normal call clearing.
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
VoiceTxDuration=375300 ms
The following is sample output from the show call history voice brief command:
Router# show call history voice brief
<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
sig:<on/off> <codec> (payload size)
Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
The following is sample output from the show call history fax command:
Router# show call history fax
DisconnectText=normal call clearing.: Normal connection
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
DisconnectText=normal call clearing.
ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]
The following is sample output from the show call history fax brief command:
Router# show call history fax brief
<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
2 : 5996450hs.25 +-1 +3802 pid:100 Answer 408
tx:0/0 rx:0/0 1F (T30 T1 EOM timeout)
Telephony : tx:38020/38020/0ms g729r8 noise:0dBm acom:0dBm
2 : 5996752hs.26 +-1 +3500 pid:110 Originate uut1@linux2.allegro.com
tx:0/0 rx:0/0 3F (The e-mail was not sent correctly. Remote SMTP server said: 354 )
IP 14.0.0.1 AcceptedMime:0 DiscardedMime:0
3 : 6447851hs.27 +1111 +3616 pid:310 Originate 576341.
tx:11/14419 rx:0/0 10 (Normal connection)
Telephony : tx:36160/11110/25050ms g729r8 noise:115dBm acom:-14dBm
3 : 6447780hs.28 +1182 +4516 pid:0 Answer
tx:0/0 rx:0/0 10 (normal call clearing.)
IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0
4 : 6464816hs.29 +1050 +3555 pid:310 Originate 576341.
tx:11/14413 rx:0/0 10 (Normal connection)
Telephony : tx:35550/10500/25050ms g729r8 noise:115dBm acom:-14dBm
4 : 6464748hs.30 +1118 +4517 pid:0 Answer
tx:0/0 rx:0/0 10 (normal call clearing.)
IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0
5 : 6507900hs.31 +1158 +2392 pid:100 Answer 4085763413
tx:0/0 rx:3/3224 10 (Normal connection)
Telephony : tx:23920/11580/12340ms g729r8 noise:0dBm acom:0dBm
5 : 6508152hs.32 +1727 +2140 pid:110 Originate uut1@linux2.allegro.com
tx:0/2754 rx:0/0 3F (service or option not available, unspecified)
IP 14.0.0.4 AcceptedMime:0 DiscardedMime:0
6 : 6517176hs.33 +1079 +3571 pid:310 Originate 576341.
tx:11/14447 rx:0/0 10 (Normal connection)
Telephony : tx:35710/10790/24920ms g729r8 noise:115dBm acom:-14dBm
6 : 6517106hs.34 +1149 +4517 pid:0 Answer
tx:0/0 rx:0/0 10 (normal call clearing.)
IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0
7 : 6567382hs.35 +1054 +3550 pid:310 Originate 576341.
tx:11/14411 rx:0/0 10 (Normal connection)
Telephony : tx:35500/10540/24960ms g729r8 noise:115dBm acom:-14dBm
7 : 6567308hs.36 +1128 +4517 pid:0 Answer
tx:0/0 rx:0/0 10 (normal call clearing.)
IP 0.0.0.0 AcceptedMime:0 DiscardedMime:0
Table 28 provides an alphabetical listing of the fields displayed in the output from the show call history command and a description of each field.
Table 28 show call history Field Descriptions
Field
|
Description
|
ACOMLevel
|
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.
|
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.
|
CallOrigin
|
Call origin: answer or originate.
|
ChargedUnits
|
Total number of charging units applying to this peer since system startup. The unit of measure for this field is hundredths of a 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 the gateway call.
|
ConnectTime
|
Time at which this 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.
|
DisconnectCause
|
Description explaining why this call was disconnected.
|
DisconnectText
|
Descriptive text explaining the reason for the disconnect.
|
DisconnectTime
|
Time when this call was disconnected.
|
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.
|
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 redundancy parameters available because voice data was lost or not received in time from the voice gateway for this call.
|
GapFillWithSilence
|
Duration of a voice signal replaced with silence because voice data was lost or not received in time for this call.
|
GapFillWithPrediction
|
Duration of a 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.
|
HiWaterPlayoutDelay
|
High-water mark Voice Playout FIFO Delay during this voice call.
|
Index
|
Dial peer identification number.
|
InfoType
|
Information type for this call; for example, voice or fax.
|
Last Buffer Drain/Fill Event
|
Time since the last jitter buffer drain or fill event, in seconds.
|
LogicalIfIndex
|
Index number of the logical voice port for this call.
|
LoWaterPlayoutDelay
|
Low-water mark Voice Playout FIFO Delay during this voice call.
|
Modem passthrough signaling method is nse
|
Indicates that this is a modem pass-through call and named signaling events (NSEs)—also called 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
|
Average 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.
|
Percent Packet Loss
|
Total percent packet loss.
|
PeerAddress
|
Destination pattern or number associated with this peer.
|
PeerId
|
ID value of the peer entry table 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 where 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.
|
ReceivePackets
|
Number of packets received by this peer during this call.
|
RemoteIPAddress
|
Remote system IP address for this call.
|
RemoteUDPPort
|
Remote system 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 RSVP QoS for this call.
|
Session Protocol
|
Session protocol used for an Internet call between the local and remote router through the IP backbone.
|
Session Target
|
Session target of the peer used for this call.
|
SetUpTime
|
Value of the system UpTime 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).
|
Time between Buffer Drain/Fills
|
Minimum and maximum durations between jitter buffer drain or fill events, in seconds.
|
TransmitBytes
|
Number of bytes sent by this peer during this call.
|
TransmitPackets
|
Number of packets sent by this peer during this call.
|
TxDuration
|
Duration of the transmit path open from this peer to the voice gateway for this call.
|
VAD
|
Specifies 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.
|
Related Commands
Command
|
Description
|
show call active
|
Displays the active call information for voice calls or fax transmissions in progress.
|
show dial-peer voice
|
Displays configuration information for dial peers.
|
show num-exp
|
Displays how the number expansions are configured in Voice over IP.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show call history video record
To display information about video calls, use the show call history video record command in privileged EXEC mode.
show call history video record
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
The command introduced in Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
Use this command to review statistics about recent incoming and outgoing video calls.
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays information about two video calls:
Router# show call history video record
CallDuration = 39006 seconds
DisconnectText = remote hangup
Remote NSAP = 47.0091810000000002F26D4901.00107B09C645.C8
Local NSAP = 47.0091810000000002F26D4901.00107B4832E1.C8
vcd = 414, vpi = 0, vci = 158
VideoSlot = 1, VideoPort = 0
CallDuration = 557 seconds
DisconnectText = local hangup
Remote NSAP = 47.0091810000000002F26D4901.00107B09C645.C8
Local NSAP = 47.0091810000000002F26D4901.00107B4832E1.C8
vcd = 364, vpi = 0, vci = 108
VideoSlot = 1, VideoPort = 0
show call history voice record
To display Call Detail Record (CDR) events in the call history table, use the show call history voice record command in privileged EXEC mode.
show call history voice record
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
The command introduced in Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Examples
The following example displays a sample of voice call history records showing a local call between two telephones attached to the same Cisco MC3810 multiservice concentrator:
Router# show call history voice record
ConnectionId=[0x2C7AEFDC 0x59830001 0x0 0xB0AAA3]
Media=TELE, TxDuration= 1418 ms
ConnectTime=1158046 x 10ms
DisconectTime=1158188 x 10ms
DisconnectText=local onhook
ConnectionId=[0x2C7AEFDC 0x59830001 0x0 0xB0AAA3]
Media=TELE, TxDuration= 1422 ms
ConnectTime=1158046 x 10ms
DisconectTime=1158188 x 10ms
DisconnectText=remote onhook
Table 29 describes the significant fields shown in the display.
Table 29 show call history voice record Field Descriptions
Field
|
Description
|
ConnectionID
|
Global call identifier for this voice call.
|
Media
|
Medium over which the call is carried. If the call is carried over the (telephone) access side, the entry will be TELE. If the call is carried over the voice network side, the entry will be either ATM, FR (for Frame Relay), or HDLC.
|
LowerIFName
|
Physical lower interface information. Appears only if the medium is either ATM, FR, or HDLC.
|
TxDuration
|
The length of the call. Appears only if the medium is TELE.
|
CalledNumber
|
The called number.
|
CallingNumber
|
The calling number.
|
SetupTime
|
Time the call setup started.
|
ConnectTime
|
Time the call is connected.
|
DisconnectTime
|
Time the call is disconnected.
|
DisconnectText
|
Descriptive text explaining the reason for the disconnect.
|
Related Commands
Command
|
Description
|
show call active voice
|
Displays the Voice over IP active call table.
|
show dial-peer voice
|
Displays configuration information for dial peers.
|
show num-exp
|
Displays how the number expansions are configured in Voice over IP.
|
show voice port
|
Displays configuration information about a specific voice port.
|
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.
show call resource voice stats
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
This command displays the H.323 resources that are monitored when the resource threshold command is used to configure and enable resource threshold reporting.
Examples
The following example shows the resource statistics for an H.323 gateway:
Router# show call resource voice stats
Resource Monitor - Dial-up Resource Statistics Information:
Table 30 describes the significant fields shown in the display.
Table 30 show call resource voice stats Field Descriptions
Statistic
|
Definition
|
Total channels
|
Number of channels physically configured for the resource.
|
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.
|
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.
|
Free channels
|
Number of addressable channels that are free.
|
Pending channels
|
Number of addressable channels that are pending in loadware download.
|
Disabled channels
|
Number of addressable channels that are physically down or that have been disabled administratively with the shutdown or busyout command.
|
Related Commands
Command
|
Description
|
resource threshold
|
Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.
|
show call resource voice threshold
|
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 show call resource voice threshold command in privileged EXEC mode.
show call resource voice threshold
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced on the Cisco AS5300 univeral access server.
|
Usage Guidelines
This command displays the H.323 resource thresholds that are configured with the resource threshold command.
Examples
The following example 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:
Threshold State: low_threshold_hit
Related Commands
Command
|
Description
|
resource threshold
|
Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.
|
show call resource voice stats
|
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 show call rsvp-sync conf command in privileged EXEC mode.
show call rsvp-sync conf
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(3)XI1
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers, the Cisco MC3810 multiservice concentrator, and on the Cisco AS5300 and Cisco AS5800 universal access servers.
|
12.1(5)T
|
This command was integrated into Cisco IOS Release 12.1(5)T.
|
Examples
The following example shows sample output from the show call rsvp-sync conf 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
Table 31 describes the significant fields shown in the display
Table 31 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
|
call rsvp-sync
|
Enables synchronization between RSVP and the H.323 voice signaling protocol.
|
call rsvp-sync resv-timer
|
Sets the timer for RSVP reservation setup.
|
debug call rsvp-sync events
|
Displays the events that occur during RSVP synchronization.
|
show call rsvp-sync stats
|
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 show call rsvp-sync stats command in privileged EXEC mode.
show call rsvp-sync stats
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
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.
|
Examples
The following example shows sample output from the show call rsvp-sync stats 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
Table 32 describes the significant fields shown in the display.
Table 32 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
|
call rsvp-sync
|
Enables synchronization between RSVP and the H.323 voice signaling protocol.
|
call rsvp-sync resv-timer
|
Sets the timer for RSVP reservation setup.
|
debug call rsvp-sync events
|
Displays the events that occur during RSVP synchronization.
|
show call rsvp-sync conf
|
Displays the RSVP synchronization configuration.
|
show cdapi
To display the Call Distributor Application Programming Interface (CDAPI), use the show cdapi command in privileged EXEC mode.
show cdapi
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
CDAPI is the internal application programming interface (API) that provides an interface between signaling stacks and applications.
Examples
The following is output for the show cdapi command:
Registered CDAPI Applications/Stacks
====================================
Application TSP CDAPI Application
Application Type(s) Voice Facility Signaling
Call ID = 0x39, Call Type = VOICE, Application = TSP CDAPI Application
Used Msg Buffers 0, Free Msg Buffers 1600
Used Raw Buffers 1, Free Raw Buffers 799
Used Large-Raw Buffers 0, Free Large-Raw Buffers 80
Related Commands
Command
|
Description
|
isdn protocol-emulate
|
Configures the Layer 2 and Layer 3 port protocol of a BRI voice port or a PRI interface to emulate NT (network) or TE (user) functionality.
|
isdn switch type
|
Configures the Cisco AS5300 universal access server PRI interface to support Q.SIG signaling.
|
pri-group nec-fusion
|
Configures your NEC PBX to support FCCS.
|
show rawmsg
|
Displays the raw messages owned by the required component.
|
show ces clock-select
To display the setting of the network clock for the specified port, use the show ces clock-select command in privileged EXEC mode.
show ces slot/port clock-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 router.
|
Examples
The following is sample output from the show ces clock-select 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
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, enter the show connect command in privileged EXEC mode.
show connect {all | elements | name | id | port {T1 | E1} slot/port}}
Syntax Description
all
|
Displays a table of all configured connections.
|
elements
|
Displays registered hardware or software interworking elements.
|
name
|
Displays 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
|
Displays the status of 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 show connect all command displays these IDs.
|
port
|
Displays the status of a connection that you specify by indicating the type of controller (T1 or E1) and location of the interface.
|
T1
|
Specifies a T1 controller.
|
E1
|
Specifies an E1 controller.
|
slot/port
|
The location of the T1 or E1 controller port whose connection status you want to see. Valid values for slot and port are 0 and 1. The slash must be entered.
|
Defaults
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 routers.
|
12.0(7)T
|
The command introduced in Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
This command shows drop-and-insert connections on the Cisco 2600 and 3600 series.
The command 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:
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
TDM timeslots in use: 14-18 (5 total)
TDM timeslots in use: 14-18
Internal Switching Elements: VIC TDM Switch
Router# show connect name Test
TDM timeslots in use: 1-13 (13 total)
TDM timeslots in use: 1-13
Internal Switching Elements: VIC TDM Switch
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 show controllers rs366 command in privileged EXEC mode.
show controllers rs366 slot port
Syntax Description
slot
|
Slot location of the VDM module. On the Cisco MC3810 multiservice concentrator, this value is either 1 or 2. If you do not enter the correct location, the command is rejected.
|
port
|
Port location of the EIA/TIA-366 interface in the VDM module. On the Cisco MC3810 multiservice concentrator, this value is 0.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
The command introduced in Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays information about the RS-366 controller:
Router# show controller 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
Table 33 describes the significant fields shown in the display.
Table 33 show controllers 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 show the channel-associated signaling (CAS) and ISDN PRI state in detail, use the show controllers timeslots command in privileged EXEC mode.
show controllers t1/e1 controller-number timeslots timeslot-range
Syntax Description
tl/e1
|
Specifies the type of interface.
|
controller-number
|
Specifies the controller number of CAS or ISDN PRI time slot. Range 0 through 7.
|
timeslots
|
Displays DS0 information.
|
timeslot-range
|
Specifies time slot range 1 through 31 for E1, 1 through 24 for T1.
|
Defaults
No default
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
|
Support for Cisco AS5400 universal access servers was added.
|
Usage Guidelines
Use the show controllers t1/e1 timeslots command to display the CAS and ISDN PRI channel state in detail. This command shows whether the DS0 channels of a controller are in idle, in-service, maintenance, or busyout states. Enter the show controllers t1/e1 command to display statistics about the T1 or E1 links.
Examples
The following example shows that the CAS state is enabled on the Cisco AS5300 universal access server with a T1 PRI card:
Router# show controllers timeslots
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
The following example shows that the ISDN PRI state is enabled on the Cisco AS5300 universal access server with a T1 PRI card:
DS0 Type Modem <-> Service Channel Rx Tx
State State A B C D A B C D
---------------------------------------------------------------------------
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
show controllers voice
To display information about voice-related hardware, use the show controllers voice command in privileged EXEC mode.
Syntax Description
This command has no arguments or keywords.
Defaults
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.
Examples
The following is an example of the output from the show controllers voice command:
Router# show controllers voice
STDA 0xFF STDB 0xFF SARA 0xAD SARB 0xFF SAXA 0xFF SAXB 0x0 STCR 0x3F
STAR 0x65 OMDR 0xE2 VNSR 0x0 PMOD 0x4C PBNR 0xFF POFD 0xF0 POFU 0x18
PCSR 0x1 PICM 0x0 CMD1 0xA0 CMD2 0x70 CBNR 0xFF CTAR 0x2 CBSR 0x20 CSCR
DSP 0 Host Port Interface:
HPI Control Register 0x202
InterfaceStatus 0x2A MaxMessageSize 0x80
RxRingBufferSize 0x6 TxRingBufferSize 0x9
pInsertRx 0x4 pRemoveRx 0x4 pInsertTx 0x6 pRemoveTx 0x6
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 4AC7 5F08 91D1 0000 0000 7DF1 69E5 63E1 63E2
0020: 6E7C ED67 DE5D DB5C DC60 EC7E 6BE1 58D3 50CD 4DCE
0040: 50D2 5AE5 7868 DA52 CE4A C746 C647 C94B D25A EAF4
0060: 5DD7 4FCD 4ACA 4ACC 4FD3 5DE8 F769 DC58 D352 D253
0080: D65B E573 6CDF 59D3 4ECF 4FD0
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 1CDD 3E48 3B74 0000 0000 3437 3D4C F0C8 BBB5
0020: B2B3 B7BF D25B 4138 3331 3339 435F CFBD B6B2 B1B4
0040: BBC8 7E48 3B34 3131 363D 4FDE C3B9 B3B1 B3B8 C2DB
0060: 533F 3833 3235 3B48 71CC BDB7 B4B5 B8BF CF67 483D
0080: 3836 383C 455B DAC6 BDB9 B9BB
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 4AC8 5F08 9221 0000 0000 54DA 61F5 EF60 DA53
0020: CF4F CD4E D256 DB63 FCEE 5FDA 55D1 50CF 4FD3 56D8
0040: 5DE1 6E7C EC60 DC59 D655 D456 D85D DF6A F4F4 69E2
0060: 5CDD 5BDC 5BDE 61E9 6DF1 FF76 F16D E96A E566 EA6A
0080: EB6F F16D EF79 F776 F5F5 73F0
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 1CDE 3E48 3BC4 0000 0000 C0CC EC54 453E 3C3C
0020: 3F47 56F3 D1C7 C1BF C0C6 CEE1 6752 4A46 4648 4E59
0040: 6FE4 D6CF CDCE D2DA E57E 675E 5B5B 5E62 6B76 FCF6
0060: F6FA 7D75 7373 7BF5 EAE1 DCDA DADD E6FE 6559 514D
0080: 4D4E 5563 EFD9 CDC8 C5C6 CAD1
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 4AC6 5F08 9181 0000 0000 DD5B DC5E E161 E468
0020: FAFD 6CE1 5AD3 53D1 53D7 61EC EA59 CF4A C644 C344
0040: CA4E D86C 60D0 48C2 3EBD 3CBD 3EC0 47CF 5976 DF4F
0060: C945 C242 C146 C94E D668 73DB 54CE 4DCC 4DCE 53DB
0080: 64F9 ED63 DC59 DA58 DC5D E46C
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 1CDC 3E48 3B24 0000 0000 5B5B 5D62 6A76 FCF5
0020: F5F9 7D78 7374 7CF5 EAE1 DDDA DBDD E7FE 6559 514E
0040: 4D4F 5663 EFD8 CDC8 C6C6 CAD1 E760 4E46 403F 4047
0060: 5173 D5C7 BFBC BCBE C5D4 6D4C 3F3B 3939 3D46 5ADB
0080: C5BC B7B6 B8BD C8E8 4F3F 3835
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 4AC6 5F08 9181 0000 003C DD5B DC5E E161 E468
0020: FAFD 6CE1 5AD3 53D1 53D7 61EC EA59 CF4A C644 C344
0040: CA4E D86C 60D0 48C2 3EBD 3CBD 3EC0 47CF 5976 DF4F
0060: C945 C242 C146 C94E D668 73DB 54CE 4DCC 4DCE 53DB
0080: 64F9 ED63 DC59 DA58 DC5D E46C
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 1CDC 3E48 3B24 0000 003C 5B5B 5D62 6A76 FCF5
0020: F5F9 7D78 7374 7CF5 EAE1 DDDA DBDD E7FE 6559 514E
0040: 4D4F 5663 EFD8 CDC8 C6C6 CAD1 E760 4E46 403F 4047
0060: 5173 D5C7 BFBC BCBE C5D4 6D4C 3F3B 3939 3D46 5ADB
0080: C5BC B7B6 B8BD C8E8 4F3F 3835
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 4AC7 5F08 91D1 0000 003C 7DF1 69E5 63E1 63E2
0020: 6E7C ED67 DE5D DB5C DC60 EC7E 6BE1 58D3 50CD 4DCE
0040: 50D2 5AE5 7868 DA52 CE4A C746 C647 C94B D25A EAF4
0060: 5DD7 4FCD 4ACA 4ACC 4FD3 5DE8 F769 DC58 D352 D253
0080: D65B E573 6CDF 59D3 4ECF 4FD0
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 1CDD 3E48 3B74 0000 003C 3437 3D4C F0C8 BBB5
0020: B2B3 B7BF D25B 4138 3331 3339 435F CFBD B6B2 B1B4
0040: BBC8 7E48 3B34 3131 363D 4FDE C3B9 B3B1 B3B8 C2DB
0060: 533F 3833 3235 3B48 71CC BDB7 B4B5 B8BF CF67 483D
0080: 3836 383C 455B DAC6 BDB9 B9BB
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 4AC8 5F08 9221 0000 003C 54DA 61F5 EF60 DA53
0020: CF4F CD4E D256 DB63 FCEE 5FDA 55D1 50CF 4FD3 56D8
0040: 5DE1 6E7C EC60 DC59 D655 D456 D85D DF6A F4F4 69E2
0060: 5CDD 5BDC 5BDE 61E9 6DF1 FF76 F16D E96A E566 EA6A
0080: EB6F F16D EF79 F776 F5F5 73F0
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 1CDE 3E48 3BC4 0000 003C C0CC EC54 453E 3C3C
0020: 3F47 56F3 D1C7 C1BF C0C6 CEE1 6752 4A46 4648 4E59
0040: 6FE4 D6CF CDCE D2DA E57E 675E 5B5B 5E62 6B76 FCF6
0060: F6FA 7D75 7373 7BF5 EAE1 DCDA DADD E6FE 6559 514D
0080: 4D4E 5563 EFD9 CDC8 C5C6 CAD1
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 1CDA 3E48 3A84 0000 003C E75F 4E46 403F 4147
0020: 5174 D5C7 BFBC BCBE C5D4 6C4C 3F3B 3939 3D46 5BDA
0040: C5BC B7B6 B8BD C8E9 4F3F 3834 3437 3D4C EEC8 BBB5
0060: B2B3 B8BF D35A 4138 3331 3339 435F CEBD B6B1 B1B4
0080: BBC9 7C48 3B34 3131 363D 4FDE
packet_length 100 channel_id 1 packet_id 0 process id 0x1
0000: 0000 4AC5 5F08 9131 0000 003C 66DE 66EB 67EE FE6E
0020: F7E7 6B68 E068 EE6A DF5C DF62 EDF1 6FF2 7A78 67DC
0040: 5EDF 62E7 64E6 66E0 7071 EA69 F86E E260 DE5D E665
0060: EB75 F0FB 6DE9 64E4 69E3 66EA 67E9 6DF9 F177 EC6E
0080: EB6E F876 F875 7D6E E966 E05D
packet_length 100 channel_id 2 packet_id 0 process id 0x1
0000: 0000 1CDB 3E48 3AD4 0000 003C C2B9 B3B1 B3B8 C2DC
0020: 523F 3733 3235 3C49 72CB BDB7 B4B5 B8BF CF67 483C
0040: 3836 373C 455C DAC6 BDB9 B9BB C0CC EE54 453E 3C3C
0060: 3F47 56F1 D1C7 C1BF C0C6 CEE1 6651 4A46 4648 4D59
0080: 70E3 D6CF CDCE D2D9 E67E 675E
Application firmware 3.1.8, Built by claux on Thu Jun 17 11:00:05 1999
VIC Interface Foreign Exchange Station 0/0, DSP instance (0x19543C0)
Singalling channel num 128 Signalling proxy 0x0 Signaling dsp 0x19543C0
tx outstanding 0, max tx outstanding 32
ptr 0x0, length 0x0, max length 0x0
dsp_number 0, Channel ID 1
received 0 packets, 0 bytes, 0 gaint packets
0 drops, 0 no buffers, 0 input errors 0 input overruns
650070 bytes output, 4976 frames output, 0 output errors, 0 output
VIC Interface Foreign Exchange Station 0/1, DSP instance (0x1954604)
Singalling channel num 129 Signalling proxy 0x0 Signaling dsp 0x1954604
tx outstanding 0, max tx outstanding 32
ptr 0x0, length 0x0, max length 0x0
dsp_number 0, Channel ID 2
received 0 packets, 0 bytes, 0 gaint packets
0 drops, 0 no buffers, 0 input errors 0 input overruns
393976 bytes output, 3982 frames output, 0 output errors, 0 output
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays configuration information and call statistics for dial peers.
|
show interface dspfarm
|
Displays hardware informatio,n including DRAM, SRAM, and the revision-level information on the line card.
|
show voice dsp
|
Displays the current status of all DSP voice channels on the Cisco MC3810 multiservice concentrator.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show csm
To display the call switching module (CSM) statistics for a particular digital signal processor (DSP) channel or all DSP channels or for a specific modem or DSP channel, use the show csm command in privileged EXEC mode.
Cisco AS5300 Universal Access Server
show csm {modem [slot/port | modem-group-number] | voice [slot/dspm/dsp/dsp-channel]}
Cisco AS5800 Universal Access Server
show csm voice [shelf/slot/port]
Syntax Description
modem
|
Specifies CSM call statistics for modems.
|
voice
|
Specifies CSM call statistics for DSP channels.
|
slot/port
|
(Optional) Specifies the location (and thereby the identity) of a specific modem.
|
modem-group-number
|
(Optional) Displays configuration for the dial peer identified by the argument modem-group-number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.
|
slot/dspm/dsp/dsp-channel
|
(Optional) Identifies the location of a particular DSP channel.
|
shelf/slot/port
|
(Optional) Identifies the location of the voice interface card.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3 NA
|
This command was introduced.
|
12.0(3)T
|
Port-specific values for the Cisco AS5300 universal access server were added.
|
12.0(7)T
|
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 show csm modem command to display the CSM call statistic information 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-number argument 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 show csm voice command to display CSM statistics for a particular DSP channel. If the slot/dspm/dsp/dsp-channel or shelf/slot/port argument is specified, the CSM call statistics for calls using the identified DSP channel will be displayed. If no argument is specified, all CSM call statistics for all DSP channels will be displayed.
Examples
The following is sample output from the Cisco AS5300 universal access server for the show csm voice command:
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
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 following is sample output from the Cisco AS5800 for the show csm voice command:
Router# show csm voice 1/8/19
VDEV_INFO:slot 8, port 19
vdev_status(0x00000401):VDEV_STATUS_ACTIVE_CALL.VDEV_STATUS_HASLOCK.
csm_state(0x00000406)=CSM_OC6_CONNECTED, csm_event_proc=0x60868B8C, current
invalid_event_count=0, wdt_timeout_count=0
watchdog timer is not activated
pri_chnl=(T1 1/0/0:22), vdev_chnl=(s8, c19)
start_chan_p=0, chan_p=62436D58, call_id=0x800D, bchan_num=22
The calling party phone number =
The called party phone number = 7511
ring_no_answer=0, ic_failure=0, ic_complete=0
dial_failure=0, oc_failure=0, oc_complete=1
oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0
remote_link_disc=0, busyout=0, modem_reset=0
call_duration_started=3d16h, call_duration_ended=00:00:00,
total_call_duration=00:00:00
Table 34 describes the significant fields shown in the display.
Table 34 show csm voice Field Descriptions
Field
|
Description
|
slot
|
Slot where the VFC resides.
|
shelf/slot/port
|
Specifies the T1 or E1 controller.
|
dspm/dsp/dsp channel
|
Indicates which DSP channel is engaged in this call.
|
dsp
|
Indicates the 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
|
Indicates 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
|
The 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
|
Indicates whether the watchdog timer is activated for this call.
|
wait_for_dialing:
|
Indicates whether this (outgoing) call is waiting for a free digit collector to become available to dial out the outgoing digits.
|
wait_for_bchan:
|
Indicates whether this (outgoing) call is waiting for a B channel to send the call out on.
|
pri_chnl=
|
Indicates which type of TDM stream is used for the PRI connection. For PRI and CAS calls, it will always be TDM_PRI_STREAM.
|
tdm_chnl=
|
Indicates 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 will always be 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=
|
Indicates the start of this call.
|
call_duration_ended=
|
Indicates the end of this call.
|
total_call_duration=
|
Indicates the 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
|
show call active voice
|
Displays the contents of the active call table.
|
show call history voice
|
Displays the contents of the call history table.
|
show num-exp
|
Displays how number expansions are configured.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show dial-peer video
To display dial-peer configuration, use the show dial-peer video command in privileged EXEC mode.
show dial-peer video [number] [summary]
Syntax Description
number
|
(Optional) A specific video dial peer. This option displays configuration information for a single dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.
|
summary
|
(Optional) Displays a summary of all video dial peer information.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
The command introduced in Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
Use this command to review video dial peer configuration.
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays detailed information about all configured video dial peers:
Router# show dial-peer video
type = videocodec, destination-pattern = 111
port signal = 1/0, port media = Serial1
nsap = 47.0091810000000050E201B101.00107B09C6F2.C8
type = videoatm, destination-pattern = 222
session-target = ATM0 svc nsap 47.0091810000000050E201B101.00E01E92ADC2.C8
type = videoatm, destination-pattern = 333
session-target = ATM0 pvc 70/70
show dial-peer voice
To display configuration information for dial peers, use the show dial-peer voice command in privileged EXEC mode.
show dial-peer voice [number] [summary]
Syntax Description
number
|
(Optional) A specific dial peer. This option displays configuration information for a single dial peer identified by the number argument. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.
|
summary
|
(Optional) Displays a summary of all voice dial peers.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
The summary keyword was added for the Cisco MC3810 multiservice concentrator.
|
12.0(3)XG
|
This command was modified to support Voice over Frame Relay (VoFR) for the Cisco 2600 series and Cisco 3600 series routers.
|
12.0(4)T
|
Support was added for VoFR for the Cisco 7200 series routers.
|
12.1(3)T
|
This command was modified for Modem Passthrough over Voice over IP on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use the show dial-peer voice privileged EXEC command to display the configuration for all Voice over IP (VoIP) and plain old telephone service (POTS) dial peers configured for the router. To show configuration information for only one specific dial peer, use the argument number to identify the dial peer.
Examples
The following is sample output from the show dial-peer voice command for a POTS dial peer:
Router# show dial-peer voice 1
tag = 1, dest-pat = `+14085291000',
group = 0, Admin state is up, Operation state is down
type = pots, prefix = `',
session-target = `', voice port =
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
The following is sample output from the show dial-peer voice command for a VoIP dial peer:
Router# show dial-peer voice 10
incall-number = `+14087',
group = 0, Admin state is up, Operation state is down
type = voip, session-target = `',
sess-proto = cisco, req-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Table 35 provides an alphabetical listing of the show dial-peer voice output fields and a description of each field.
Table 35 show dial-peer voice Field Descriptions
Field
|
Description
|
Accepted Calls
|
Number of calls accepted from this peer since system startup.
|
acc-qos
|
Lowest acceptable quality of service configured for calls for this peer.
|
Admin state
|
Administrative state of this peer.
|
answer-address
|
Answer address configured for this dial peer.
|
Charged Units
|
Total number of charging units applying to this peer since system startup. The unit of measure for this field in hundredths of a second.
|
codec
|
Default voice coder rate of speech for this peer.
|
Connect Time
|
Accumulated connect time to the peer since system startup for both incoming and outgoing calls. The unit of measure for this field is in hundredths of a second.
|
dest-pat
|
Destination pattern (telephone number) for this peer.
|
DTMF Relay
|
Indicates whether or not dual-tone multifrequency (DTMF) relay has been enabled, by using the dtmf-relay command, for this dial peer.
|
Expect factor
|
User-requested Expectation Factor of voice quality for calls through this peer.
|
fax-rate
|
Fax transmission rate configured for this peer.
|
Failed Calls
|
Number of failed call attempts to this peer since system startup.
|
group
|
Group number associated with this peer.
|
huntstop
|
Indicates whether dial-peer hunting has been turned on, by using the huntstop command, for this dial peer.
|
Icpif
|
Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer.
|
incall-number
|
Full E.164 telephone number to be used to identify the dial peer.
|
incoming called-number
|
Indicates the incoming called number if it has been set by using the incoming-called number command.
|
information type
|
Information type for this call; for example, voice or fax.
|
Last Disconnect Cause
|
Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.
|
Last Disconnect Text
|
ASCII text describing the reason for the last call termination.
|
Last Setup Time
|
Value of the System Up Time when the last call to this peer was started.
|
Modem passthrough
|
Modem pass-through signaling method is named signaling event (NSE).
|
Operation state
|
Operational state of this peer.
|
Payload type
|
NSE payload type.
|
Permission
|
Configured permission level for this peer.
|
Poor QOV Trap
|
Whether Poor Quality of Voice trap messages have been enabled or disabled.
|
Redundancy
|
Packet redundancy (RFC 2198) for modem traffic.
|
Refused Calls
|
Number of calls from this peer refused since system startup.
|
req-qos
|
Configured requested quality of service for calls for this dial peer.
|
session-target
|
Session target of this peer.
|
sess-proto
|
Session protocol to be used for Internet calls between local and remote routers through the IP backbone.
|
Successful Calls
|
Number of completed calls to this peer.
|
tag
|
Unique dial peer ID number.
|
VAD
|
Whether voice activation detection (VAD) is enabled for this dial peer.
|
Related Commands
Command
|
Description
|
show call active voice
|
Displays the Voice over IP active call table.
|
show call history voice
|
Displays the Voice over IP call history table.
|
show num-exp
|
Displays how the number expansions are configured in Voice over IP.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show dialplan incall number
To show which plain old telephone service (POTS) dial peer is matched for a specific calling number or voice port, use the show dialplan incall number command in privileged EXEC mode.
show dialplan incall voice-port number calling-number [timeout]
Syntax Description
voice-port
|
Specifies the voice port location. The syntax of this argument is platform-specific. For information on the syntax for a particular platform, see the voice-port global configuration command.
|
calling-number
|
Specifies the calling number or ANI of the incoming voice call.
|
timeout
|
(Optional) Allows matching for variable-length destination patterns.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
12.2(8)T
|
The timeout keyword was added.
|
Usage Guidelines
Use the show dialplan incall number command as a troubleshooting tool to determine which POTS dial peer is matched for an incoming call, for the selected calling number and voice port. When using the show dialplan incall number command, the router attempts to match these items in the order listed:
1.
Calling number with answer-address configured in dial peer
2.
Calling number with destination-pattern configured in dial peer
3.
Voice port with voice port configured in dial peer
The router first attempts to match a dial peer based on the calling number (ANI). If the router is unable to match a dial peer based on the calling number, it matches the call to a POTS dial peer based on the selected voice interface. If more than one dial peer uses the same voice port, the router selects the first matching dial peer. Use the timeout keyword to enable matching variable-length destination patters associated with dial peers. This can increase you r chances of finding a match for the dial peer number you specify.
Note
For actual voice calls coming into the router, the router attempts to match the called number (the dialed number identification service [DNIS] number) with the incoming called-number configured in a dial peer. The router, however, does not consider the called number when using the show dialplan incall number command.
Examples
The following example shows that an incoming call from interface 1/0/0:D with a calling number of 12345 is matched to POTS dial peer 10:
Router# show dialplan incall 1/0/0:D number 12345
information type = voice,
tag = 10, destination-pattern = `123..',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 10, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
in bound application associated: DEFAULT
out bound application associated:
incoming COR list:maximum capability
outgoing COR list:minimum requirement
type = pots, prefix = `',
session-target = `', voice-port = `1/0/0:D',
direct-inward-dial = disabled,
register E.164 number with GK = TRUE
Connect Time = 0, Charged Units = 0,
register E.164 number with GK = TRUE
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0,
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
The following example shows that if no dial peer has a destination pattern or answer address that matches the calling number of 888, the incoming call is matched to POTS dial peer 99, because the call comes in on voice port 1/0/1:D, which is the voice port configured for this dial peer:
Router# show dialplan incall 1/0/1:D number 888
information type = voice,
tag = 99, destination-pattern = `99...',
answer-address = `', preference=1,
numbering Type = `national'
group = 99, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
in bound application associated: DEFAULT
out bound application associated:
incoming COR list:maximum capability
outgoing COR list:minimum requirement
type = pots, prefix = `5',
session-target = `', voice-port = `1/0/1:D',
direct-inward-dial = enabled,
register E.164 number with GK = TRUE
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0,
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Table 36 describes the significant fields shown in the display.
Table 36 show dialplan number Field Descriptions
Field
|
Description
|
Macro Exp.
|
Expected destination pattern for this dial peer.
|
VoiceEncapPeer
|
Dial peer associated with the calling number entered.
|
tag
|
Unique number identifying the dial peer.
|
destination-pattern
|
Destination pattern (telephone number) configured for this dial peer.
|
answer-address
|
Answer address (calling number) configured for this dial peer.
|
preference
|
Hunt group preference order set for this dial peer.
|
Admin state
|
Describes the administrative state of this dial peer.
|
Operation state
|
Describes the operational state of this dial peer.
|
incoming called-number
|
Called number (DNIS) configured for this dial peer.
|
DTMF Relay
|
Whether the dtmf-relay command is enabled or disabled for this dial peer.
|
huntstop
|
Whether the huntstop command is enabled or disabled for this dial peer.
|
in bound application associated
|
The IVR application that is associated with this dial peer when this dial peer is used for an inbound call leg.
|
out bound application associated
|
The IVR application that is associated with this dial peer when this dial peer is used for an outbound call leg.
|
type
|
Type of dial peer (POTS or VoIP).
|
prefix
|
The prefix number that is added to the front of the dial string before it is forwarded to the telephony device.
|
forward-digits
|
Which digits are forwarded to the telephony interface as configured using the forward-digits command.
|
session-target
|
Displays the configured session target (IP address or host name) for this dial peer.
|
voice-port
|
The voice port through which calls come into this dial peer.
|
direct-inward-dial
|
Whether the direct-inward-dial command is enabled or disabled for this dial peer.
|
digit_strip
|
Whether digit stripping is enabled or disabled in the dial peer. Enabled is the default.
|
session-protocol
|
Session protocol to be used for Internet calls between local and remote router via the IP backbone.
|
Connect Time
|
Unit of measure indicating the call connection time associated with this dial peer.
|
Charged Units
|
Number of call units charged to this dial peer.
|
Successful Calls
|
Number of completed calls to this peer since system startup.
|
Failed Calls
|
Number of uncompleted (failed) calls to this peer since system startup.
|
Accepted Calls
|
Number of calls from this peer accepted since system startup.
|
Refused Calls
|
Number of calls from this peer refused since system startup.
|
Last Disconnect Cause
|
Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.
|
Last Disconnect Text
|
ASCII text describing the reason for the last call termination.
|
Last Setup Time
|
Value of the System Up Time when the last call to this peer was started.
|
Matched
|
Destination pattern matched for this dial peer.
|
Target
|
Matched session target (IP address or host name) for this dial peer.
|
Related Commands
Command
|
Description
|
show dial peer voice
|
Displays the configuration information for dial peers.
|
show dialplan number
|
Displays which dial peer is matched for a particular telephone number.
|
show dialplan number
To show which dial peer is reached when a particular telephone number is dialed, use the show dialplan number command in privileged EXEC mode.
show dialplan number dial string [huntstop] [timeout]
Syntax Description
dial string
|
Specifies a particular destination pattern (telephone number).
|
huntstop
|
(Optional) Terminates further dial-peer hunting upon encountering the first dial string match.
|
timeout
|
(Optional) Allows matching for variable-length destination patterns.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
12.2(1)
|
The huntstop keyword was added.
|
12.2(8)T
|
The timeout keyword was added.
|
Usage Guidelines
The show dialplan number command is used to test whether the dial plan configuration is valid and working as expected. Use the timeout keyword to enable matching variable-length destination patters associated with dial peers. This can increase you r chances of finding a match for the dial peer number you specify.
Examples
The following example shows sample output from the show dialplan number command using the destination pattern of 1001:
Router# show dialplan number 1001
information type = voice,
tag = 1003, destination-pattern = `1001',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 1003, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
type = pots, prefix = `',
session-target = `', voice-port = `1/1',
direct-inward-dial = disabled,
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
information type = voice,
tag = 1004, destination-pattern = `1001',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 1004, Admin state is up, Operation state is up,
information type = voice,
tag = 1002, destination-pattern = `1001',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 1002, Admin state is up, Operation state is up,
information type = voice,
tag = 1001, destination-pattern = `1001',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 1001, Admin state is up, Operation state is up,
The following example shows sample output from the show dialplan number command using the destination pattern of 1001 and the huntstop keyword:
Router# show dialplan number 1001 huntstop
information type = voice,
tag = 1003, destination-pattern = `1001',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 1003, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
type = pots, prefix = `',
session-target = `', voice-port = `1/1',
direct-inward-dial = disabled,
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Table 37 explains the significant fields shown in this example.
Table 37 show dialplan number Field Descriptions
Field
|
Description
|
Macro Exp.
|
Expected destination pattern for this dial peer.
|
VoiceEncapPeer
|
Dial peer associated with the destination pattern entered.
|
type
|
Type of dial peer (POTS or VoIP).
|
tag
|
Unique dial peer identifying number.
|
destination-pattern
|
Destination pattern (telephone number) configured for this dial peer.
|
answer-address
|
Answer address configured for this dial peer.
|
Admin state
|
Administrative state of this dial peer.
|
Operation state
|
Operational state of the dial peer.
|
session-target
|
Configured session target (IP address or host name) for this dial peer.
|
Connect Time
|
Unit of measure indicating the call connection time associated with this dial peer.
|
Charged Units
|
Number of call units charged to this dial peer.
|
Successful Calls
|
Number of completed calls to this peer since system startup.
|
Failed Calls
|
Number of uncompleted (failed) calls to this peer since system startup.
|
Accepted Calls
|
Number of calls accepted from this peer since system startup.
|
Refused Calls
|
Number of calls refused from this peer since system startup.
|
Last Disconnect Cause
|
Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.
|
Last Disconnect Text
|
ASCII text describing the reason for the last call termination.
|
Last Setup Time
|
Value of the System Up Time when the last call to this peer was started.
|
Matched
|
Destination pattern matched for this dial peer.
|
Target
|
Matched session target (IP address or host name) for this dial peer.
|
Related Commands
Command
|
Description
|
show dialplan incall number
|
Pairs different voice ports and telephone numbers together for troubleshooting Voice over IP.
|
show frame-relay vofr
To display information about the FRF.11 subchannels being used on Voice over Frame Relay (VoFR) data link connection identifiers (DLCIs), use the show frame-relay vofr command in privileged EXEC mode.
show frame-relay vofr [interface [dlci [cid]]]
Syntax Description
interface
|
(Optional) The specific interface type and number for which you wish to display FRF.11 subchannel information.
|
dlci
|
(Optional) The specific data link connection identifier for which you wish to display FRF.11 subchannel information.
|
cid
|
(Optional) The specific subchannel for which you wish to display information.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(4)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
This command was integrated in Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
If this command is entered without a specified interface, FRF.11 subchannel information will be displayed for all VoFR interfaces and DLCIs configured on the router.
Note
This command is currently not supported on the Cisco MC3810 multiservice concentrator for PVCs configured with the vofr cisco command or the frame-relay interface-dlci voice-encap command.
Examples
The following is sample output from the show frame-relay vofr command when an interface is not specified:
Router# show frame-relay vofr
interface vofr-type dlci cid cid-type
Serial0/0.1 VoFR 16 4 data
Serial0/0.1 VoFR 16 5 call-control
Serial0/0.1 VoFR 16 10 voice
Serial0/1.1 VoFR cisco 17 4 data
The following is sample output from the show frame-relay vofr command when an interface is specified:
Router# show frame-relay vofr serial0
interface vofr-type dlci cid cid-type
Serial0 VoFR 16 5 call-control
The following is sample output from the show frame-relay vofr command when an interface and a DLCI are specified:
Router# show frame-relay vofr serial0 16
VoFR Configuration for interface Serial0
dlci vofr-type cid cid-type input-pkts output-pkts dropped-pkts
16 VoFR 5 call-control 85982 86099 0
16 VoFR 10 voice 2172293 6370815 0
The following is sample output from the show frame-relay vofr command when an interface, a DLCI, and a CID are specified:
Router# show frame-relay vofr serial0 16 10
VoFR Configuration for interface Serial0 dlci 16
vofr-type VoFR cid 10 cid-type voice
input-pkts 2172293 output-pkts 6370815 dropped-pkts 0
Table 38 describes the significant fields shown in the display.
Table 38 show frame-relay vofr Field Descriptions
Field
|
Description
|
interface
|
Number of the interface that has been selected for observation of FRF.11 subchannels.
|
vofr-type
|
Type of VoFR DLCI being observed.
|
cid
|
The portion of the specified DLCI that is carrying the designated traffic type. A DLCI can be subdivided into 255 subchannels.
|
cid-type
|
Type of traffic carried on this subchannel.
|
input-pkts
|
Number of packets received by this subchannel.
|
output-pkts
|
Number of packets sent on this subchannel.
|
dropped-pkts
|
Total number of packets discarded by this subchannel.
|
Related Commands
Command
|
Description
|
show call active voice
|
Displays the contents of the active call table.
|
show call history voice
|
Displays the contents of the call history table.
|
show dial-peer voice
|
Displays configuration information and call statistics for dial peers.
|
show frame-relay fragment
|
Displays Frame Relay fragmentation details.
|
show frame-relay pvc
|
Displays statistics about PVCs for Frame Relay interfaces.
|
show voice-port
|
Displays configuration information about a specific voice port.
|
show gatekeeper calls
To show the status of each ongoing call of which a gatekeeper is aware, use the show gatekeeper calls command in privileged EXEC mode.
show gatekeeper calls
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(3)T
|
The command introduced in Cisco IOS Release 11.3(2)NA was integrated into Cisco IOS Release 12.0(3)T.
|
12.0(5)T
|
The output for this command was changed.
|
Usage Guidelines
Use the show gatekeeper calls command to show all active calls currently being handled by a particular MCM gatekeeper. If you have forced a disconnect for either a particular call or all calls associated with a particular MCM gatekeeper by using the clear h323 gatekeeper call command, the system will not display information about those calls.
Examples
The following is sample output from the show gatekeeper calls command:
Router# show gatekeeper calls
Total number of active calls = 1.
Endpt(s):Alias E.164Addr CallSignalAddr Port RASSignalAddr Port
src EP:epA 90.0.0.11 1720 90.0.0.11 1700
src PX:pxA 90.0.0.01 1720 90.0.0.01 24999
dst PX:pxB 172.21.139.90 1720 172.21.139.90 24999
Table 39 describes the significant fields shown in the display.
Table 39 show gatekeeper calls Field Descriptions
Field
|
Description
|
LocalCallID
|
Identification number of the call.
|
Age(secs)
|
The age of the call in seconds.
|
BW(Kbps)
|
The bandwidth in use, in kilobits per second.
|
Endpoint(s)
|
Lists the role of each endpoint (terminal, gateway, or proxy) in the call (originator, target, or proxy), and the call signaling and registration, admission, and status (RAS) protocol address.
|
Alias
|
H.323-ID or Email-ID of the endpoint.
|
E.164Addr
|
E.164 address of the endpoint.
|
CallSignalAddr
|
Call signaling IP address of the endpoint.
|
Port
|
Call signaling port number of the endpoint.
|
RASSignalAddr
|
RAS IP address of the endpoint.
|
Port
|
RAS port number of the endpoint.
|
Related Commands
Command
|
Description
|
clear h323 gateway call
|
Forces a specific call or all calls currently active on the gatekeeper to disconnect.
|
show gatekeeper endpoints
To display the status of all registered endpoints for a gatekeeper, use the show gatekeeper endpoints command in EXEC mode.
show gatekeeper endpoints
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(5)T
|
The display format was modified for H.323 Version 2.
|
Usage Guidelines
Use this command to display the status of all registered endpoints for a gatekeeper.
Examples
The following is sample output from the show gatekeeper endpoints command:
Router# show gatekeeper endpoints
CallsignalAddr Port RASSignalAddr Port Zone Name Type F
--------------- ---- ------------- ----- ---------- ----- --
172.21.127.8 1720 172.21.127.8 24999 sj-gk MCU
H323-ID:joe@cisco.com
172.21.13.88 1720 172.21.13.88 1719 sj-gk VOIP-GW O H323-ID:la-gw
Table 40 describes the significant fields shown in the display.
Table 40 show gatekeeper endpoints Field Descriptions
Field
|
Description
|
CallsignalAddr
|
Call signaling IP address of the endpoint. If the endpoint is also registered with alias, a list of all aliases registered for that endpoint should be listed on the line below.
|
Port
|
Call signaling port number of the endpoint.
|
RASSignalAddr
|
Registration, admission, and status (RAS) protocol IP address of the endpoint.
|
Port
|
RAS port number of the endpoint.
|
Zone Name
|
Zone name (gatekeeper ID) that this endpoint registered in.
|
Type
|
The endpoint type (for example, terminal, gateway, or MCU).
|
F
|
S—Indicates that the endpoint is statically entered from the alias command rather than being dynamically registered through RAS messages. O—Indicates that the endpoint, which is a gateway, has sent notification that it is nearly out of resources.
|
Related Commands
Command
|
Description
|
show gatekeeper gw-type-prefix
|
Displays the gateway technology prefix table.
|
show gatekeeper zone status
|
Displays the status of zones related to a gatekeeper.
|
show gateway
|
Displays the current gateway status.
|
show gatekeeper gw-type-prefix
To display the gateway technology prefix table, use the show gatekeeper gw-type-prefix command in privileged EXEC mode.
show gatekeeper gw-type-prefix
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(5)T
|
The display format was modified for H.323 Version 2.
|
Usage Guidelines
Use the show gatekeeper gw-type-prefix command to display the gateway technology prefix table.
Examples
The following is sample output for a gatekeeper that is controlling two local zones, sj-gk and la-gk:
Router# show gatekeeper gw-type-prefix
GATEWAY TYPE PREFIX TABLE
===========================
Prefix:12#* (Default gateway-technology)
Zone sj-gk master gateway list:
172.21.13.22:1720 sj-gw2 (out-of-resources)
Zone sj-gk prefix 408....... priority gateway list(s):
172.21.13.22:1720 sj-gw2 (out-of-resources)
Prefix:7#* (Hopoff zone la-gk)
Statically-configured gateways (not necessarily currently registered):
Zone la-gk master gateway list:
171.69.127.11:1720 la-gw1
171.69.127.22:1720 la-gw2
Table 41 describes the fields contained in the show gatekeeper gw-type-prefix sample output.
Table 41 show gatekeeper gw-type-prefix Field Descriptions
Field
|
Description
|
Prefix
|
The technology prefix defined with the gw-type-prefix command.
|
Zone sj-gk master gateway list
|
A list of all the gateways registered to zone sj-gk with the technology prefix, under which they are listed. (This display shows that gateways sj-gw1, sj-gw2, and sj-gw3 have registered in zone sj-gk with the technology prefix 12#.)
|
Zone sj-gk prefix 408....... priority gateway list(s)
|
A list of prioritized gateways to handle calls to area code 408.
|
Priority 10
|
Highest priority level. Gateways listed following "Priority 10" are given the highest priority when selecting a gateway to service calls to the specified area code. (In this display, gateway sj-gw1 is given the highest priority to handle calls to the 408 area code.)
|
Priority 5
|
Any gateway that does not have a priority level assigned to it defaults to priority 5.
|
(out-of-resources)
|
Indication that the displayed gateway has sent a "low-in-resources" notification.
|
(Hopoff zone la-gk)
|
Any call specifying this technology prefix should be directed to hop off in the la-gk zone, no matter what the area code of the called number is. (In this display, calls specifying technology prefix 7# are always routed to zone la-gk, regardless of the actual zone prefix in the destination address.)
|
Zone la-gk master gateway list
|
A list of all the gateways registered to la-gk with the technology prefix under which they are listed. (This display shows that gateways la-gw1 and la-gw2 have registered in zone la-gk with the technology prefix 7#. No priority lists are displayed here because none were defined for zone la-gk.)
|
(Default gateway-technology)
|
If no gateway-type prefix is specified in a called number, then gateways registering with 12# are the default type to be used for the call.
|
(Statically-configured gateways)
|
Lists all IP addresses and port numbers of gateways that are incapable of supplying technology-prefix information when they register. This display shows that when gateways 1.1.1.1:1720 and 2.2.2.2:1720 register, they will be considered to be of type 7#.
|
Related Commands
Command
|
Description
|
show gatekeeper calls
|
Displays the status of each ongoing call that a gatekeeper is aware of.
|
show gatekeeper endpoints
|
Displays the status of all registered endpoints for a gatekeeper.
|
show gateway
|
Displays the current gateway status.
|
show gatekeeper servers
To see a list of currently registered and statically configured triggers on this gatekeeper router, enter the show gatekeeper servers command in EXEC mode.
show gatekeeper servers [gkid]
Syntax Description
gkid
|
(Optional) The local gatekeeper name to which this trigger applies.
|
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2500 series, Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
Enter this command to show all server triggers (whether dynamically registered from the external servers or statically configured from the command line interface) on this gatekeeper. If gkid is specified, only triggers applied to the specified gatekeeper zone appear. If the gkid argument is not specified, server triggers for all local gatekeeper zones on this router appear.
Examples
This example shows the operating information of the specified gk102 server:
Router# show gatekeeper servers gk102
GATEKEEPER SERVERS STATUS
=========================
Gatekeeper Server listening port:20000
Server IP address:1.14.93.28:42387
Server type:dynamically registered
Server IP address:1.14.93.43:3820
Server type:CLI-configured
Connection Status:inactive
Server IP address:1.14.93.28:42387
Server type:dynamically registered
Destination Info:M:nilkant@zone14.com
Destination Info:E:1800.......
Redirect Reason:Call forwarded no reply
Redirect Reason:Call deflection
Related Commands
Command
|
Description
|
debug gatekeeper server
|
Traces all the message exchanges between the Cisco IOS gatekeeper and the external applications. Shows any errors that occur in sending messages to the external applications or in parsing messages from the external applications.
|
show gatekeeper status
To show overall gatekeeper status, including authorization and authentication status, zone status, and so on, use the show gatekeeper status command in EXEC mode.
show gatekeeper status
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(3)T
|
The command introduced in Cisco IOS Release 11.3(2)NA was integrated into Cisco IOS Release 12.0(3)T.
|
Examples
The following is sample output from the show gatekeeper status command:
Router# show gatekeeper status
Zone Name: gk-px4.cisco.com
Table 42 describes the significant fields shown in the display.
Table 42 show gatekeeper status Field Descriptions
Field
|
Description
|
Gatekeeper State
|
Gatekeeper status has the following values:
• UP is operational.
• DOWN is administratively shut down.
• INACTIVE is administratively enabled, that is, the no shutdown command has been issued, but no local zones have been configured.
• HSRP STANDBY indicates that the gatekeeper is on hot standby and will take over when the currently active gatekeeper fails.
|
Zone Name
|
Zone name.
|
Accounting
|
Authorization and accounting status.
|
Security
|
Security status.
|
show gatekeeper zone prefix
To display the zone prefix table, use the show gatekeeper zone prefix command in privileged EXEC mode.
show gatekeeper zone prefix
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
Examples
The following is an example of output from the show gatekeeper zone prefix command:
Router# show gatekeeper zone prefix
Table 43 describes the significant fields shown in the display.
Table 43 show gatekeeper zone prefix Field Descriptions
Field
|
Description
|
GK-NAME
|
Gatekeeper name.
|
E164-PREFIX
|
The E.164 prefix and a dot that acts as a wildcard for matching each remaining number in the telephone number.
|
show gatekeeper zone status
To display the status of zones related to a gatekeeper, use the show gatekeeper zone status command in privileged EXEC mode.
show gatekeeper zone status
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers.
|
12.0(5)T
|
This display format was modified for H.323 Version 2.
|
Usage Guidelines
Use this command to display the status of all zones related to a gatekeeper.
Examples
The following is an example of output from the show gatekeeper zone status command:
Router# show gatekeeper zone status
GK name Domain Name RAS Address PORT FLAGS MAX-BW CUR-BW
------- ----------- ----------- ---- ----- ------ ------
sj.xyz.com xyz.com 1.14.93.85 1719 LS 0
All Other Subnets :(Enabled)
PROXY USAGE CONFIGURATION :
inbound Calls from germany.xyz.com :
to terminals in local zone sj.xyz.com :use proxy
to gateways in local zone sj.xyz.com :do not use proxy
Outbound Calls to germany.xyz.com
from terminals in local zone germany.xyz.com :use proxy
from gateways in local zone germany.xyz.com :do not use proxy
Inbound Calls from all other zones :
to terminals in local zone sj.xyz.com :use proxy
to gateways in local zone sj.xyz.com :do not use proxy
Outbound Calls to all other zones :
from terminals in local zone sj.xyz.com :do not use proxy
from gateways in local zone sj.xyz.com :do not use proxy
tokyo.xyz.co xyz.com 172.21.139.89 1719 RS 0
milan.xyz.co xyz.com 171.69.57.90 1719 RS 0
Table 44 describes the significant fields shown in the display.
Table 44 show gatekeeper zone status Field Descriptions
Field
|
Description
|
GK name
|
The gatekeeper name (also known as zone name), which is truncated after 12 characters in the display.
|
Domain Name
|
The domain with which the gatekeeper is associated.
|
RAS Address
|
The registration, admission, and status (RAS) protocol address of the gatekeeper.
|
FLAGS
|
Displays the following information:
• S = static (CLI-configured, not DNS-discovered)
• L = local
• R = remote
|
MAX-BW
|
The maximum bandwidth for the zone, in kbps.
|
CUR-BW
|
The current bandwidth in use, in kbps.
|
SUBNET ATTRIBUTES
|
A list of subnets controlled by the local gatekeeper.
|
PROXY USAGE CONFIGURATION
|
Inbound and outbound proxy policies as configured for the local gatekeeper (or zone).
|
Related Commands
Command
|
Description
|
show gatekeeper calls
|
Displays the status of each ongoing call of which a gatekeeper is aware.
|
show gatekeeper endpoints
|
Displays the status of all registered endpoints for a gatekeeper.
|
show gateway
|
Displays the current gateway status.
|
show gateway
To display the current gateway status, use the show gateway command in privileged EXEC mode.
show gateway
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(6)NA2
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers.
|
12.0(5)T
|
This display format was modified for H.323 V2.
|
Usage Guidelines
This command displays the current gateway status.
Examples
The following example shows the report that appears when the gateway is not registered with a gatekeeper:
Gateway gateway1 is not registered to any gatekeeper
H323 resource thresholding is Enabled but NOT Active
H323 resource threshold values:
DSP: Low threshold 60, High threshold 70
DS0: Low threshold 60, High threshold 70
This following example indicates that an E.164 address has been assigned to the gateway:
Gateway gateway1 is registered to Gatekeeper gk1
The following example shows the report that appears when the gateway is registered with a gatekeeper and H.323 resource threshold reporting is enabled with the resource threshold command:
Gateway gateway1 is registered to Gatekeeper gk1
H323 resource thresholding is Enabled and Active
H323 resource threshold values:
DSP: Low threshold 60, High threshold 70
DS0: Low threshold 60, High threshold 70
The following example shows the report that appears when the gateway is registered with a gatekeeper and H.323 resource threshold reporting is disabled with the no resource threshold command:
Gateway gateway1 is registered to Gatekeeper gk1
H323 resource thresholding is Disabled
Related Commands
Command
|
Description
|
resource threshold
|
Configures a gateway to report H.323 resource availability to the gatekeeper of the gateway.
|
show interface dspfarm
To display digital signal processor (DSP) information on the two-port T1/E1 high-density port adapter for the Cisco 7200 series, use the show interface dspfarm command in privileged EXEC mode.
show interface dspfarm [slot/port]
Syntax Description
slot
|
(Optional) Slot location of the port adapter.
|
/port
|
(Optional) Port number on the port adapter.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XE
|
This command was introduced on the Cisco 7200 series router.
|
12.1(1)T
|
This command was integrated into the Cisco IOS 12.1(1)T.
|
Usage Guidelines
The local time-division multiplexing (TDM) cross-connect map can be displayed.
Examples
The following example is sample output from the show interface dspfarm command for port adapter slot 0 of chassis slot 3, on the Cisco 7200 series router:
Router# show interface dspfarm 3/0
DSPfarm3/0 is up, line protocol is up
MTU 256 bytes, BW 12000 Kbit, DLY 0 usec,
reliability 255/255, txload 4/255, rxload 1/255
Encapsulation VOICE, loopback not set
C549 DSP Firmware Version:MajorRelease.MinorRelease (BuildNumber)
DSP Boot Loader:255.255 (255)
Medium Complexity Application:3.2 (5)
High Complexity Application:3.2 (5)
Total DSPs 30, DSP0-DSP29, Jukebox DSP id 30
Total sig channels 120 used 24, total voice channels 120 used 0
0 active calls, 0 max active calls, 0 total calls
30887 rx packets, 0 rx drops, 30921 tx packets, 0 tx frags
0 curr_dsp_tx_queued, 29 max_dsp_tx_queued
Last input never, output never, output hang never
Last clearing of "show interface" counters never
Output queue 0/0, 0 drops; input queue 0/75, 0 drops
5 minute input rate 13000 bits/sec, 94 packets/sec
5 minute output rate 193000 bits/sec, 94 packets/sec
30887 packets input, 616516 bytes, 0 no buffer
Received 0 broadcasts, 0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort
30921 packets output, 7868892 bytes, 0 underruns
0 output errors, 0 collisions, 0 interface resets
0 output buffer failures, 0 output buffers swapped out
Table 45 describes the significant fields shown in the display.
Table 45 show interface dspfarm Field Descriptions
Field
|
Description
|
DSPfarm3/0 is up
|
DSPfarm interface is operating. The interface state can be up, down, or administratively down.
|
Line protocol is
|
Indicates whether the software processes that handle the line protocol consider the line usable or if it has been taken down by an administrator.
|
Hardware
|
Version number of the hardware.
|
MTU
|
256 bytes.
|
BW
|
12000 Kbit.
|
DLY
|
Delay of the interface in microseconds.
|
Reliability
|
Reliability of the interface as a fraction of 255 (255/255 is 100% reliability, calculated as an expediential average over 5 minutes).
|
Txload
|
Number of packets sent.
|
Rxload
|
Number of packets received.
|
Encapsulation
|
Encapsulation method assigned to the interface.
|
Loopback
|
Loopback conditions.
|
C549 DSP Firmware Version
|
The version of DSP firmware installed.
|
DSP Boot Loader
|
DSP boot loader version.
|
DSP Application
|
DSP application code version.
|
Medium Complexity Application
|
DSP Medium Complexity Application code version.
|
High Complexity Application
|
DSP High Complexity Application code version.
|
Total DSPs
|
Total DSPs that are equipped in the PA.
|
DSP0-DSP
|
DSP number range.
|
Jukebox DSP id
|
Jukebox DSP number.
|
Down DSPs
|
DSPs not in service.
|
Total sig channels...used...
|
Total number of signal channels used.
|
Total voice channels...used...
|
Total number of voice channels used.
|
Active calls
|
Number of active calls.
|
Max active calls
|
Maximum number of active calls.
|
Total calls
|
Total number of calls.
|
Rx packets
|
Number of received (rx) packets.
|
Rx drops
|
Number of rx packets dropped at PA.
|
Tx packets
|
Number of transmit (tx) packets.
|
Tx frags
|
Number of tx packets that were fragmented.
|
Curr_dsp_tx_queued
|
Number of tx packets that are being queued at host DSP queues.
|
Max_dsp_tx_queued
|
The max total tx packets that were queued at host DSP queues.
|
Last input
|
Number of hours, minutes, and seconds since the last packet was successfully received by an interface. Useful for knowing when a dead interface failed. This counter is updated only when packets are process switched and not when packets are fast switched.
|
Output
|
Number of hours, minutes, and seconds since the last packet was successfully sent by the interface. Useful for knowing when a dead interface failed. This counter is updated only when packets are process switched and not when packets are fast switched.
|
Output hang
|
Number of hours, minutes, and seconds (or never) since the interface was last reset because of a transmission that took too long. When the number of hours in any of the "last" fields exceeds 24 hours, the number of days and hours is printed. If that field overflows, asterisks (**) are printed.
|
Last clearing of "show interface" counters
|
Number of times the "show interface" counters were cleared.
|
queueing strategy
|
First-in, first-out queueing strategy (other queueing strategies you might see are priority-list, custom-list, and weighted fair).
|
Output queue
|
Number of packets in output queue.
|
Drops
|
Number of packets dropped because of a full queue.
|
Input queue
|
Number of packets in input queue.
|
Minute input rate
|
Average number of bits and packets received per minute in the past 5 minutes.
|
Bits/sec
|
Average number of bits sent per second.
|
Packets/sec
|
Average number of packets sent per second.
|
Packets input
|
Total number of error-free packets received by the system.
|
Bytes
|
Total number of bytes, including data and MAC encapsulation, in the error free packets received by the system.
|
No buffer
|
Number of received packets discarded because there was no buffer space in the main system. Compare with ignored count. Broadcast storms on Ethernets and bursts of noise on serial lines are often responsible for no-input-buffer events.
|
Received...broadcasts
|
Total number of broadcast or multicast packets received by the interface.
|
Runts
|
Number of packets that are discarded because they are smaller than the minimum packet size for the medium. For instance, any Ethernet packet that is less than 64 bytes is considered a runt.
|
Giants
|
Number of packets that are discarded because they exceed the maximum packet size for the medium. For instance, any Ethernet packet that is greater than 1518 bytes is considered a giant.
|
Throttles
|
Number of times the receiver on the port was disabled, possibly because of buffer or processor overload.
|
Input errors
|
Number of packet input errors.
|
CRC
|
Cyclic redundancy checksum generated by the originating LAN station or far end device does not match the checksum calculated from the data received. On a LAN, this usually indicates noise or transmission problems on the LAN interface or the LAN bus itself. A high number of CRCs is usually the result of collisions or a station sending bad data. On a serial link, CRCs usually indicate noise, gain hits, or other transmission problems on the data link.
|
Frame
|
Number of packets received incorrectly having a CRC error and a noninteger number of octets. On a serial line, this is usually the result of noise or other transmission problems.
|
Overrun
|
Number of times the serial receiver hardware was unable to hand received data to a hardware buffer because the input rate exceeded the ability of the receiver to handle the data.
|
Ignore
|
Number of received packets ignored by the interface because the interface hardware ran low on internal buffers. These buffers are different from the system buffers mentioned previously in the buffer description. Broadcast storms and bursts of noise can cause the ignored count to be incremented.
|
Abort
|
Illegal sequence of one bits on the interface.
|
Packets output
|
Total number of messages sent by the system.
|
Bytes
|
Total number of bytes, including data and MAC encapsulation, sent by the system.
|
Underruns
|
Number of times that the far end transmitter has been running faster than the near-end router's receiver can handle.
|
Output errors
|
Sum of all errors that prevented the final transmission of datagrams out of the interface being examined. Note that this value might not balance with the sum of the enumerated output errors; some datagrams can have more than one error, and others can have errors that do not fall into any of the specifically tabulated categories.
|
Collisions
|
Number of messages re-sent because of an Ethernet collision. Collisions are usually the result of an overextended LAN (Ethernet or transceiver cable too long, more than two repeaters between stations, or too many cascaded multiport transceivers). A packet that collides is counted only once in output packets.
|
Interface resets
|
Number of times an interface has been completely reset. Resetting can happen if packets queued for transmission were not sent within a certain interval. If the system notices that the carrier detect line of an interface is up, but the line protocol is down, it periodically resets the interface in an effort to restart it. Interface resets can also occur when an unrecoverable interface processor error occurs, or when an interface is looped back or shut down.
|
Output buffer failures
|
Number of failed buffers.
|
Output buffers swapped out
|
Number of buffers swapped out.
|
show mgcp
To display Media Gateway Control Protocol (MGCP) configuration information, use the show mgcp command in EXEC mode.
show mgcp
Defaults
No defaults
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced for the Cisco AS5300 universal access server.
|
12.1(3)T
|
Output was updated to show additional gateway and platform information.
|
Examples
The following displays an example of the command format and output for show mgcp.
MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE
MGCP call-agent: 192.168.10.10 2302 Initial protocol service is MGCP
mgcp block-newcalls DISABLED
MGCP dtmf-relay disabled for all codec types
MGCP request timeout 500, MGCP request retries 3
MGCP gateway port: 2427, MGCP maximum waiting delay 3000
MGCP restart delay 5, MGCP vad DISABLED
MGCP sdp simple DISABLED, MGCP cisco fgdos DISABLED
MGCP codec type g711ulaw, MGCP packetization period 20
MGCP JB threshold lwm 30, MGCP JB threshold hwm 150
MGCP LAT threshold lwm 150, MGCP LAT threshold hwm 300
MGCP PL threshold lwm 1000, MGCP PL threshold hwm 10000
MGCP playout mode is adaptive 60, 4, 200 in msec
MGCP IP ToS low delay disabled, MGCP IP ToS high throughput disabled
MGCP IP ToS high reliability disabled, MGCP IP ToS low cost disabled
MGCP IP precedence 3, MGCP default package: trunk-package
MGCP supported packages: gm-package dtmf-package trunk-package rtp-package
as-packagescript-package
Table 46 describes the significant fields shown in the display.
Table 46 show mgcp Field Descriptions
MGCP Admin State...Oper State
|
The administrative and operational state of the MGCP daemon. The administrative state controls starting and stopping the application using the mgcp and mgcp block-newcalls commands. The operational state controls normal MGCP operations.
|
MGCP call-agent
|
The address of the call agent specified in the mgcp command.
|
Initial protocol service is...
|
Indicates the protocol initiated for this session.
|
MGCP block-newcalls enabled
|
The state of the mgcp block-newcalls command.
|
MGCP dtmf-relay
|
The setting for the mgcp dtmf-relay command.
|
MGCP modem passthru
|
Indicates whether a call agent will be involved in relaying modem data.
|
MGCP request timeout
|
The setting for the mgcp request timeout command.
|
MGCP request retries
|
The setting for the mgcp request retries command.
|
MGCP gateway port
|
The UDP port specification.
|
MGCP maximum waiting delay
|
The setting for the mgcp max-waiting-delay command.
|
MGCP restart delay
|
The setting for the mgcp restart-delay command.
|
MGCP vad
|
The setting for the mgcp vad command.
|
MGCP sdp simple
|
Indicates whether the simple sdp protocol is being used.
|
MGCP cisco fgdos
|
For Cisco use only.
|
MGCP codec type
|
The setting for the mgcp codec command.
|
MGCP packetization period
|
The packetization period parameter setting for the mgcp codec command.
|
MGCP JB threshold lwm
|
The jitter buffer minimum threshold parameter setting for the mgcp quality-threshold command.
|
MGCP JB threshold hwm
|
The jitter buffer maximum threshold parameter setting for the mgcp quality-threshold command.
|
MGCP LAT threshold lwm
|
The latency minimum threshold parameter setting for the mgcp quality-threshold command.
|
MGCP LAT threshold hwm
|
The latency maximum threshold parameter setting for the mgcp quality-threshold command.
|
MGCP PL threshold lwm
|
The packet loss minimum threshold parameter setting for the mgcp quality-threshold command.
|
MGCP PL threshold hwm
|
The packet loss maximum threshold parameter setting for the mgcp quality-threshold command.
|
MGCP playout mode
|
The jitter buffer packet size type and size.
|
MGCP IP ToS low delay
|
The low-delay parameter setting for the mgcp ip-tos command.
|
MGCP IP ToS high throughput
|
The high-throughput parameter setting for the mgcp ip-tos command.
|
MGCP IP ToS high reliability
|
The high-reliability parameter setting for the mgcp ip-tos command.
|
MGCP IP ToS low cost
|
The low-cost parameter setting for the mgcp ip-tos command.
|
MGCP IP precedence
|
The precedence parameter setting for the mgcp ip-tos command.
|
MGCP default package
|
The default-package parameter setting for the mgcp default-package command.
|
MGCP supported packages
|
The packages supported in this session.
|
Related Commands
Command
|
Description
|
mgcp
|
Starts the MGCP daemon.
|
show mgcp connection
|
Displays connection-related MGCP configuration information.
|
show mgcp endpoint
|
Displays endpoint-specific MGCP configuration information.
|
show mgcp statistics
|
Displays statistical MGCP configuration information.
|
show mgcp connection
To display Media Gateway Control Protocol (MGCP) configuration information, use the show mgcp connection command in EXEC mode.
show mgcp connection
Syntax Description
connection
|
Displays the active MGCP-controlled connections.
|
Defaults
No defaults
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced for the Cisco AS5300 universal access server.
|
12.1(3)T
|
Output was updated to show additional gateway and platform information.
|
Examples
Following is an example of an MGCP configuration displaying active MGCP-controlled connections.
Router# show mgcp connection
Endpoint Call_ID(C) Conn_ID(I) (P)ort (M)ode (S)tate (C)odec (E)vent[SIFL] (R)esult[EA]
1. S0/DS1-0/1 C=103,23,24 I=0x8 P=16586,16634 M=3 S=4,4 C=5 E=2,0,0,2 R=0,0
2. S0/DS1-0/2 C=103,25,26 I=0x9 P=16634,16586 M=3 S=4,4 C=5 E=0,0,0,0 R=0,0
3. S0/DS1-0/3 C=101,15,16 I=0x4 P=16506,16544 M=3 S=4,4 C=5 E=2,0,0,2 R=0,0
4. S0/DS1-0/4 C=101,17,18 I=0x5 P=16544,16506 M=3 S=4,4 C=5 E=0,0,0,0 R=0,0
5. S0/DS1-0/5 C=102,19,20 I=0,6 P=16572,16600 M=3 S=4,4 C=5 E=2,0,0,2 R=0,0
6. S0/DS1-0/6 C=102,21,22 I=0x7 P=16600,16572 M=3 S=4,4 C=5 E=0,0,0,0 R=0,0
Total number of active calls 6
Table 47 describes the significant fields shown in the display.
Table 47 show mgcp connection Field Descriptions
Endpoint
|
The endpoint for each call shown in the digital endpoint naming convention of slot number (S0) and digital line (DS1-0) number (1).
|
Call_ID(C)
|
The MGCP call ID sent by the call agent, the internal Call Control Application Programming Interface (CCAPI) call ID for this endpoint, and the peer call legs CCAPI call ID.
(CCAPI is an API that provides call control facilities to applications.)
|
Conn_ID(I)
|
The connection ID generated by the gateway and sent in the ACK message.
|
(P)ort
|
The ports used for this connection. The first port is the local UDP port. The second port is the remote UDP port.
|
(M)ode
|
The call mode, where:
0—An invalid value for mode.
1—The gateway should only send packets.
2—The gateway should only receive packets.
3—The gateway can send and receive packets.
4—The gateway should neither send nor receive packets.
5—The gateway should place the circuit in loopback mode.
6—The gateway should place the circuit in test mode.
7—The gateway should use the circuit for network access for data.
8—The gateway should place the connection in network loopback mode.
9—The gateway should place the connection in network continuity test mode.
10— The gateway should place the connection in conference mode.
All other values are used for internal debugging.
|
(S)tate
|
The call state. The values are used for internal debugging purposes.
|
(C)odec
|
The codec identifier. The values are used for internal debugging purposes.
|
(E)vent [SIFL]
|
Used for internal debugging.
|
(R)esult [EA]
|
Used for internal debugging.
|
Related Commands
Command
|
Description
|
mgcp
|
Starts the MGCP daemon.
|
show mgcp
|
Displays general MGCP configuration information.
|
show mgcp endpoint
|
Displays endpoint-specific MGCP configuration information.
|
show mgcp statistics
|
Displays statistical MGCP configuration information.
|
show mgcp endpoint
To display Media Gateway Control Protocol (MGCP) configuration information, use the show mgcp endpoint command in EXEC mode.
show mgcp endpoint
Syntax Description
endpoint
|
Displays the MGCP-controlled endpoints.
|
Defaults
No defaults
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced for the Cisco AS5300 universal access server.
|
12.1(3)T
|
Output was updated to show additional gateway and platform information.
|
Examples
The following example shows how endpoints are configured:
Router# show mgcp endpoint
T1/0 ds0-group 0 timeslots 1-24 type none
T1/1 ds0-group 0 timeslots 1-24 type none
T1/2 ds0-group 0 timeslots 1-24 type none
T1/3 ds0-group 0 timeslots 1-24 type none
Related Commands
Command
|
Description
|
mgcp
|
Starts the MGCP daemon.
|
show mgcp
|
Displays general MGCP configuration information.
|
show mgcp connection
|
Displays connection-related MGCP configuration information.
|
show mgcp statistics
|
Displays statistical MGCP configuration information.
|
show mgcp statistics
To display Media Gateway Control Protocol (MGCP) configuration information, use the show mgcp statistics command in EXEC mode.
show mgcp statistics
Syntax Description
statistics
|
Displays MGCP statistics regarding network messages that have been received and sent.
|
Defaults
No defaults
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced for the Cisco AS5300 universal access server.
|
12.1(3)T
|
Output was updated to show additional gateway and platform information.
|
Examples
Following is an example of an MGCP configuration displaying MGCP statistics of network messages that have been received and sent.
Router# show mgcp statistics
Unrecognized rx pkts 0, MGCP message parsing errors 0
Duplicate MGCP ack tx 0, Invalid versions count 0
CreateConn rx 4, successful 0, failed 0
DeleteConn rx 2, successful 2, failed 0
ModifyConn rx 4, successful 4, failed 0
DeleteConn tx 0, successful 0, failed 0
NotifyRequest rx 0, successful 4, failed 0
AuditConnection rx 0, successful 0, failed 0
AuditEndpoint rx 0, successful 0, failed 0
RestartInProgress tx 1, successful 1, failed 0
Notify tx 0, successful 0, failed 0
IP address based Call Agents statistics:
IP address 10.24.167.3, Total msg rx 8, successful 8, failed 0
Table 48 describes the significant fields shown in the display.
Table 48 show mgcp statistics Field Descriptions
UDP pkts rx, tx
|
The number of UDP packets transmitted and received by the gateway's MGCP application from the Call Agent.
|
Unrecognized rx pkts
|
The number of unrecognized UDP packets received by the MGCP application.
|
MGCP message parsing errors
|
The number of MGCP messages received with parsing errors.
|
Duplicate MGCP ack tx messages
|
The number of duplicate MGCP acknowledgment messages transmitted to the Call Agent.
|
Invalid versions count
|
The number of MGCP messages received with invalid MGCP protocols version.
|
CreateConn rx
|
The number of Create Connection (CRCX) messages received by the gateway, the number that were successful, and the number that failed.
|
DeleteConn rx
|
The number of Delete Connection (DLCX) messages received by the gateway, the number that were successful, and the number that failed.
|
NotifyRequest rx
|
The number of Notify Request (RQNT) messages received by the gateway, the number that were successful, and the number that failed.
|
AuditConnection rx
|
The number of Audit Connection (AUCX) message received by the gateway, the number that were successful, and the number that failed.
|
AuditEndpoint rx
|
The number of Audit Endpoint (AUEP) messages received by the gateway, the number that were successful, and the number that failed.
|
RestartinProgress tx
|
The number of Restart in Progress (RSIP) messages transmitted by the gateway, the number that were successful, and the number that failed.
|
Notify tx
|
The number of Notify (NTFY) messages transmitted by the gateway, the number that were successful, and the number that failed.
|
ACK tx, NACK tx
|
The number of Acknowledgment and Negative Acknowledgment messages transmitted by the gateway.
|
ACK rx, NACK rx
|
The number of Acknowledgment and Negative Acknowledgment messages received by the gateway.
|
IP address based Call Agents statistics: IP address, Total msg rx
|
IP address of the Call Agent, the total number of MGCP messages received from that Call Agent, the number of messages that were successful, and the number of messages that failed.
|
Related Commands
Command
|
Description
|
mgcp
|
Starts the MGCP daemon.
|
show mgcp
|
Displays general MGCP configuration information.
|
show mgcp connection
|
Displays connection-related MGCP configuration information.
|
show mgcp endpoint
|
Displays endpoint-specific MGCP configuration information.
|
show num-exp
To show the number expansions configured, use the show num-exp command in privileged EXEC mode.
show num-exp [dialed-number]
Syntax Description
dialed-number
|
(Optional) Dialed number.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 platform.
|
12.0(3)T
|
This command was supported on the Cisco AS5300 universal access server platform.
|
12.0(4)XL
|
This command was supported on the Cisco AS5800 platform.
|
12.0(7)XK
|
This command was supported on the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into Cisco IOS 12.1(2)T.
|
Usage Guidelines
Use the show num-exp privileged EXEC command to display all the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
Examples
The following is sample output from the show num-exp command:
Dest Digit Pattern = '0...' Translation = '+14085270...'
Dest Digit Pattern = '1...' Translation = '+14085271...'
Dest Digit Pattern = '3..' Translation = '+140852703..'
Dest Digit Pattern = '4..' Translation = '+140852804..'
Dest Digit Pattern = '5..' Translation = '+140852805..'
Dest Digit Pattern = '6....' Translation = '+1408526....'
Dest Digit Pattern = '7....' Translation = '+1408527....'
Dest Digit Pattern = '8...' Translation = '+14085288...'
Table 49 describes the significant fields shown in the display.
Table 49 show num-exp Field Descriptions
Field
|
Description
|
Dest Digit Pattern
|
Index number identifying the destination telephone number digit pattern.
|
Translation
|
Expanded destination telephone number digit pattern.
|
Related Commands
Command
|
Description
|
show call active voice
|
Displays the Voice over IP active call table.
|
show call history voice
|
Displays the Voice over IP call history table.
|
show dial-peer voice
|
Displays configuration information for dial peers.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show pots csm
To show the current state of calls and the most recent event received by the call switching module (CSM) on the Cisco 800 series router, use the show pots csm command in EXEC mode.
show pots csm port
Syntax Description
Command Modes
EXEC
Command History
Release
|
Modification
|
12.1.(2)XF
|
This command was introduced on the Cisco 800 series routers.
|
Examples
The following is an example of show pots csm command output:
CSM Finite State Machine:
Call 0 - State: idle, Call Id: 0x0
Event: CSM_EVENT_NONE Cause: 0
Call 1 - State: idle, Call Id: 0x0
Event: CSM_EVENT_NONE Cause: 0
Call 2 - State: idle, Call Id: 0x0
Event: CSM_EVENT_NONE Cause: 0
Related Commands
Command
|
Description
|
test pots dial
|
Dial a telephone number for the POTS port on the router by using a dial application on your workstation.
|
test pots disconnect
|
Disconnect a telephone call for the POTS port on the router.
|
show pots status
To display the settings of the telephone port physical characteristics and other information on the telephone interfaces of the Cisco 800 series, use the show pots status command in privileged EXEC mode.
show pots status [1 | 2]
Syntax Description
1
|
(Optional) Display the settings of telephone port 1.
|
2
|
(Optional) Display the settings of telephone port 2.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco 800 series router.
|
Usage Guidelines
The show pots status command displays the settings and information for both telephone ports.
Examples
The following is sample output from the show pots status command.
POTS Global Configuration:
Dialing Method: Overlap, Tone Source: Remote, CallerId Support: YES
Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,
Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec
Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec
TX Gain: 6dB, RX Loss: -6dB,
Hook Switch Finite State Machine:
Hook Switch Register: 10, Suspend Poll: 0
CODEC Finite State Machine:
Connection: None, Call Type: Two Party, Direction: Rx only
Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,
Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec
Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec
TX Gain: 6dB, RX Loss: -6dB,
SPI Addr: 2, DSLAC Revision: 4
SLIC Cmd: 0D, TX TS: 00, RX TS: 00
Op Fn: 6F, Op Fn2: 00, Op Cond: 00
AISN: 6D, ELT: B5, EPG: 32 52 00 00
Z: EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 F0
B: 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01
X: AB 40 3B 9F A8 7E 22 97 36 A6 2A AE
R: 01 11 01 90 01 90 01 90 01 90 01 90
CSM Finite State Machine:
Call 0 - State: idle, Call Id: 0x0
Call 1 - State: idle, Call Id: 0x0
Call 2 - State: idle, Call Id: 0x0
Hook Switch Finite State Machine:
Hook Switch Register: 20, Suspend Poll: 0
CODEC Finite State Machine:
Connection: None, Call Type: Two Party, Direction: Rx only
Line Type: 600 ohm, PCM Encoding: u-law, Disc Type: OSI,
Ringing Frequency: 20Hz, Distinctive Ring Guard timer: 0 msec
Disconnect timer: 1000 msec, Disconnect Silence timer: 5 sec
TX Gain: 6dB, RX Loss: -6dB,
SPI Addr: 3, DSLAC Revision: 4
SLIC Cmd: 0D, TX TS: 00, RX TS: 00
Op Fn: 6F, Op Fn2: 00, Op Cond: 00
AISN: 6D, ELT: B5, EPG: 32 52 00 00
Z: EA 23 2A 35 A5 9F C2 AD 3A AE 22 46 C2 F0
B: 29 FA 8F 2A CB A9 23 92 2B 49 F5 37 1D 01
X: AB 40 3B 9F A8 7E 22 97 36 A6 2A AE
R: 01 11 01 90 01 90 01 90 01 90 01 90
CSM Finite State Machine:
Call 0 - State: idle, Call Id: 0x0
Call 1 - State: idle, Call Id: 0x0
Call 2 - State: idle, Call Id: 0x0
Table 50 describes the significant fields shown in the display.
Table 50 show pots status Field Descriptions
Field
|
Descriptions
|
POTS Global Configuration
|
Displays the settings of the telephone port physical characteristic commands. Also displays the following:
• TX GAIN—Current transmit gain of telephone ports.
• RX LOSS—Current transmit loss of telephone ports.
• Filter Mask—Value determines which filters are currently enabled or disabled in the telephone port hardware.
• Adaptive Cntrl Mask—Value determines if telephone port adaptive line impedance hardware is enabled or disabled.
|
Hook Switch Finite State Machine
|
Device driver that tracks state of telephone port hook switch.
|
CODEC Finite State Machine
|
Device driver that controls telephone port codec hardware.
|
CODEC Registers
|
Register contents of telephone port codec hardware.
|
CODEC Coefficients
|
Codec coefficients selected by telephone port driver. Selected line type determines codec coefficients.
|
CSM Finite State Machine
|
State of call-switching module (CSM) software.
|
Time Slot Control
|
Register that determines if telephone port voice or data packets are sent to an ISDN B channel.
|
Related Commands
Command
|
Description
|
pots country
|
Configures telephones, fax machines, or modems connected to a Cisco 800 series router to use country-specific default settings for each physical characteristic.
|
pots dialing-method
|
Specifies how the Cisco 800 series router collects and sends digits dialed on your connected telephones, fax machines, or modems.
|
pots disconnect-supervision
|
Specifies how a Cisco 800 series router notifies the connected telephones, fax machines, or modems when the calling party has disconnected.
|
pots disconnect-time
|
Specifies the interval in which the disconnect method is applied if telephones, fax machines, or modems connected to a Cisco 800 series router fail to detect that a calling party has disconnected.
|
pots distinctive-ring-guard-time
|
Specifies a delay in which a telephone port can be rung after a previous call is disconnected (Cisco 800 series routers).
|
pots encoding
|
Specifies the PCM encoding scheme for telephones, fax machines, or modems connected to a Cisco 800 series router.
|
pots line-type
|
Specifies the impedance of telephones, fax machines, or modems connected to a Cisco 800 series router.
|
pots ringing-freq
|
Specifies the frequency at which telephones, fax machines, or modems connected to a Cisco 800 series router ring.
|
pots silence-time
|
Specifies the interval of silence after a calling party disconnects (Cisco 800 series router).
|
pots tone-source
|
Specifies the source of dial, ringback, and busy tones for telephones, fax machines, or modems connected to a Cisco 800 series router.
|
show proxy h323 calls
To list each active call on the proxy, use the show proxy h323 calls command in privileged EXEC mode.
show proxy h323 calls
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(3)T
|
The command was integrated into Cisco IOS Release 12.0(3)T and supported on the Cisco MC3810 multiservice concentrator.
|
Examples
The following is sample output from the show proxy h323 calls command:
Router# show proxy h323 calls
Conference ID = [277B87C0A283D111B63E00609704D8EA]
Calling endpoint call signalling address = 55.0.0.41
Calling endpoint aliases:
H323_ID: ptel11@zone1.com
Call state = Media Streaming
Time call was initiated = 731146290 ms
show proxy h323 detail-call
To display the details of a particular call on a proxy, use the show proxy h323 detail-call command in privileged EXEC mode.
show proxy h323 detail-call call-key
Syntax Description
call-key
|
Specifies the call you want to display. The call-key argument is derived from the show proxy h323 calls display.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(3)T
|
The command was integrated into Cisco IOS Release 12.0(3)T and supported on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
The show proxy h323 detail-call command can be used with or without the proxy statistics enabled.
Examples
The following is sample output from the show proxy h323 detail-call command without the proxy statistics enabled:
Router# show proxy h323 detail-call 1
ConferenceID = [277B87C0A283D111B63E00609704D8EA]
Calling endpoint aliases:
H323_ID: ptel11@zone1.com
H323_ID: ptel21@zone2.com
Peer proxy call signalling address = 55.0.0.41
Time call was initiated = 731146290 ms
Call state = Media Streaming
H245 logical channels for call leg pte111@zone1.com<->px1@zone.com
Time created = 731146317 ms
Time created = 731146316 ms
Time created = 731146318 ms
Time created = 731146317 ms
H245 logical channels for call leg pte111@zone1.com<->50.0.0.41:
Time created = 731146317 ms
Time created = 731146316 ms
Time created = 731146318 ms
Time created = 731146317 ms
The following is sample output from the show proxy h323 detail-call command with the proxy statistics enabled:
Router# show proxy h323 detail-call 1
ConferenceID = [677EB106BD0D111976200002424F832]
Calling endpoint call signalling address = 172.21.127.49
Calling endpoint aliases:
H323_ID: mcs@sanjose.cisco.com
Peer proxy call signalling address = 171.68.183.199
H323_ID: proxy.sanjose.cisco.com
Time call was initiated = 730949651 ms
Call state = H245 open logical channels
H245 logical channels for call leg intel2 <-> cisco7-pxy:
RTP stream from intel2 to cisco7-pxy
Time created = 730949676 ms
RTP stream from intel2 to cisco7-pxy
Time created = 730949658 ms
RTP stream from cisco7-pxy to intel2
Time created = 730949664 ms
Packet Received Count = 3390
Packet Out of Sequence Count = 0
Number of initial packets used for Arrival-Spacing bin setup = 200
min_arrival_spacing = 0(ms) max_arrival_spacing = 856(ms)
Average Arrival Rate = 86(ms)
Arrival-Spacing(ms) Packet-Count
==============================
Min Jitter = 34(ms) Max Jitter = 408(ms)
Average Jitter Rate = 117
Jitter Rate(ms) Packet-Count
Number of initial packets used for Arrival-Spacing bin setup = 200
min_arrival_spacing = 32(ms) max_arrival_spacing = 96(ms)
Average Arrival Rate = 60(ms)
Arrival-Spacing(ms) Packet-Count
==============================
Min Jitter = 0(ms) Max Jitter = 28(ms)
Jitter Rate(ms) Packet-Count
H245 logical channels for call leg cisco7-pxy <->
RTP stream from cisco7-pxy to proxy.sanjose.cisco.com
Time created = 730949676 ms
Packet Received Count = 3398
Packet Out of Sequence Count = 0
Number of initial packets used for Arrival-Spacing bin setup = 200
min_arrival_spacing = 0(ms) max_arrival_spacing = 872(ms)
Average Arrival Rate = 85(ms)
Arrival-Spacing(ms) Packet-Count
==============================
Min Jitter = 55(ms) Max Jitter = 447(ms)
Average Jitter Rate = 127
Jitter Rate(ms) Packet-Count
RTP stream from cisco7-pxy to proxy.sanjose.cisco.com
Time created = 730949658 ms
Packet Received Count = 2537
Packet Out of Sequence Count = 0
Number of initial packets used for Arrival-Spacing bin setup = 200
min_arrival_spacing = 0(ms) max_arrival_spacing = 32716(ms)
Average Arrival Rate = 112(ms)
Arrival-Spacing(ms) Packet-Count
==============================
Min Jitter = 32(ms) Max Jitter = 1256(ms)
Average Jitter Rate = 121
Jitter Rate(ms) Packet-Count
RTP stream from proxy.sanjose.cisco.com to cisco7-pxy
Time created = 730949664 ms
RTP stream from proxy.sanjose.cisco.com to cisco7-pxy
Time created = 730949661 ms
Related Commands
Command
|
Description
|
h323 qos
|
Enables QoS on the proxy.
|
show proxy h323 status
To display the overall status of a proxy, use the show proxy h323 status command in privileged EXEC mode.
show proxy h323 status
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced.
|
12.0(3)T
|
The command was integrated into Cisco IOS Release 12.0(3)T and supported on the Cisco MC3810 multiservice concentrator.
|
Examples
The following is sample output from the show proxy h323 status command:
Router# show proxy h323 status
H.323 Proxy Mode: Enabled
Proxy interface = Serial1: UP
Application Specific Routing: Disabled
RAS Initialization: Complete
Proxy aliases configured:
Proxy aliases assigned by Gatekeeper:
Gatekeeper multicast discovery: Disabled
Gatekeeper ID: gk.zone2.com
Gatekeeper registration succeeded
Number of calls in progress: 1
show rawmsg
To show the raw messages owned by the required component, use the show rawmsg command in privileged EXEC mode.
show rawmsg {all | tsp | vtsp | ccapi | h323}
Syntax Description
all
|
All selections below.
|
tsp
|
Telephony Service Provider subsystem.
|
vtsp
|
Voice Telephony Service Provider subsystem.
|
ccapi
|
API (Application Programming Interface) used to coordinate interaction between application and call legs (telephony or IP).
|
h323
|
H.323 subsystem.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
The number displayed for show rawmsg all should be zero to indicate that there are no memory leaks.
Examples
The following example shows how to display memory leaks from the telephony service provider:
Related Commands
Command
|
Description
|
isdn protocol-emulate
|
Configures the Layer 2 and Layer 3 port protocol of a BRI voice port or a PRI interface to emulate NT (network) or TE (user) functionality.
|
isdn switch type
|
Configures the Cisco AS5300 universal access server PRI interface to support Q.SIG signaling.
|
pri-group nec-fusion
|
Configures your NEC PBX to support FCCS.
|
show cdapi
|
Displays the CDAPI.
|
show rlm group statistics
To display the network latency of the Redundant Link Manager (RLM) group, use the show rlm group statistics command in privileged EXEC mode.
show rlm group group-number statistics
Syntax Description
group-number
|
RLM group number (0 to 255).
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Examples
The following is sample output from the show rlm group group-number statistics command:
Router# show rlm group 1 statistics
last time occurred at 02:45:48.724, total transition=1
avg=00:00:00.000, max=00:00:00.000, min=00:00:00.000, latest=00:00:00.000
last time occurred at 02:42:33.724, total transition=1
avg=00:03:15.000, max=00:03:15.000, min=00:00:00.000, latest=00:03:15.000
last time occurred at 00:00:00.000, success=0(0%), failure=0
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
last time occurred at 00:00:00.000, success=0(0%), failure=0
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
last time occurred at 00:00:00.000 for totally 0 times
Server Link Group[r1-server]:
Open the link [10.1.1.1(Loopback1), 10.1.4.1]:
last time occurred at 02:43:03.724, success=1(100%), failure=0
avg=162.000s, max=162.000s, min=0.000s, latest=162.000s
Echo over link [10.1.1.1(Loopback1), 10.1.4.1]:
last time occurred at 02:47:15.724, success=91(62%), failure=54
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
Open the link [10.1.1.2(Loopback2), 10.1.4.2]:
last time occurred at 02:43:03.724, success=1(100%), failure=0
avg=162.000s, max=162.000s, min=0.000s, latest=162.000s
Echo over link [10.1.1.2(Loopback2), 10.1.4.2]:
last time occurred at 02:47:19.724, success=95(63%), failure=54
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
Server Link Group[r2-server]:
Open the link [10.1.1.1(Loopback1), 10.1.5.1]:
last time occurred at 02:46:06.724, success=0(0%), failure=1
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
Echo over link [10.1.1.1(Loopback1), 10.1.5.1]:
last time occurred at 02:47:18.724, success=0(0%), failure=85
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
Open the link [10.1.1.2(Loopback2), 10.1.5.2]:
last time occurred at 02:46:06.724, success=0(0%), failure=1
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
Echo over link [10.1.1.2(Loopback2), 10.1.5.2]:
last time occurred at 02:47:18.724, success=0(0%), failure=85
avg=0.000s, max=0.000s, min=0.000s, latest=0.000s
Table 51 describes the significant fields shown in the display.
Table 51 show rlm group statistics Field Descriptions
Field
|
Description
|
Link_up
|
Statistics collected when RLM group is in link up state.
|
total transition
|
Total number of transitions into a particular RLM group state.
|
avg
|
How long the average time interval lasts.
|
max
|
How long the maximum time interval lasts.
|
min
|
How long the minimum time interval lasts.
|
latest
|
How long the most recent time interval lasts.
|
Link_down
|
Statistics collected when RLM group is in the link down state.
|
Link_recovered
|
Statistics collected when RLM group is in the link recovery state.
|
Link_switched
|
Statistics collected when RLM group is in the link switching state.
|
Server_changed
|
Statistics collected for when and how many times RLM server failover happens.
|
Server Link Group[r1-server]
|
Statistics collected for those signaling links defined under a particular server link group, for example, r1-server.
|
Open the link
|
Statistics collected when a particular signaling link connection is open (broken).
|
Echo over link
|
Statistics collected when a particular signaling link connection is established.
|
Related Commands
Command
|
Description
|
clear interface
|
Resets the hardware logic on an interface.
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
interface
|
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
server (RLM)
|
Defines the IP addresses of the server.
|
show rlm group status
|
Displays the status of the RLM group.
|
show rlm group timer
|
Displays the current RLM group timer values.
|
shutdown (RLM)
|
Shuts down all of the links under the RLM group.
|
timer
|
Overwrites the default setting of timeout values.
|
show rlm group status
To display the status of the Redundant Link Manager (RLM) group, use the show rlm group status command in privileged EXEC mode.
show rlm group group-number status
Syntax Description
group-number
|
RLM group number (0 to 255).
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Examples
The following is sample output from the show rlm group group-number status command:
Router# show rlm group 1 status
Link State: Up Last Link Status Reported: Up
Next tx TID: 1 Last rx TID: 0
Server Link Group[r1-server]:
link [10.1.1.1(Loopback1), 10.1.4.1] = socket[active]
link [10.1.1.2(Loopback2), 10.1.4.2] = socket[standby]
Server Link Group[r2-server]:
link [10.1.1.1(Loopback1), 10.1.5.1] = socket[opening]
link [10.1.1.2(Loopback2), 10.1.5.2] = socket[opening]
Table 52 describes the significant fields shown in the display.
Table 52 show rlm group status Field Descriptions
Field
|
Description
|
User/Port
|
A list of registered RLM users and the corresponding port numbers associated with them.
|
RLM_MGR
|
RLM management module.
|
Link State
|
The current RLM group's link state for connecting to the remote end.
|
Last Link Status Reported
|
The most recent link status change is reported to RLM users.
|
Next tx TID
|
The next transaction ID for transmission.
|
Last rx TID
|
The most recent transaction ID has been received.
|
Server Link Group[r1-server]
|
The status of all signaling links configured under a particular RLM server link group r1-server.
|
socket
|
The status of the individual signaling link.
|
Server Link Group[r2-server]
|
The status of all signaling links configured under a particular RLM server link group (r2-server).
|
Related Commands
Command
|
Description
|
clear interface
|
Resets the hardware logic on an interface.
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
interface
|
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
server (RLM)
|
Defines the IP addresses of the server.
|
show rlm group status
|
Displays the status of the RLM group.
|
show rlm group timer
|
Displays the current RLM group timer values.
|
shutdown (RLM)
|
Shuts down all of the links under the RLM group.
|
timer
|
Overwrites the default setting of timeout values.
|
show rlm group timer
To display the current timer values, use the show rlm group timer command in privileged EXEC mode.
show rlm group group-number timer
Syntax Description
group-number
|
RLM group number (0 to 255).
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Examples
The following is sample output from the show rlm group group-number timer command:
Router# show rlm group 1 timer
open_wait = 3s force-down = 30s
recovery = 12s switch-link = 5s
minimum-up = 60s retransmit = 1s
Table 90 describes the significant fields shown in the display.
Table 53 show rlm group timer Field Descriptions
Field
|
Description
|
open_wait
|
Wait for the connection request to be acknowledged.
|
recovery
|
Time to allow the link to recover to backup link before declaring the link is down.
|
minimum-up
|
Minimum time to force RLM to stay in the down state to make sure the remote end detects the link state is down.
|
keepalive
|
A keepalive packet will be sent out from network access server to CSC periodically.
|
force-down
|
Minimum time to force RLM to stay in the down state to make sure that the remote end detects that the link state is down.
|
switch-link
|
The maximum transition period allows RLM to switch from a lower preference link to a higher preference link. If the switching link does not complete successfully before this timer expires, RLM will go into the recovery state.
|
retransmit
|
Because RLM is operating under UDP, it needs to resend the control packet if the packet is not acknowledged within this retransmit interval.
|
Related Commands
Command
|
Description
|
clear interface
|
Resets the hardware logic on an interface.
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
interface
|
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
server (RLM)
|
Defines the IP addresses of the server.
|
show rlm group status
|
Displays the status of the RLM group.
|
shutdown (RLM)
|
Shuts down all of the links under the RLM group.
|
timer
|
Overwrites the default setting of timeout values.
|
show rtsp client session
To display cumulative information about Real Time Streaming Protocol (RTSP) session records, use the show rtsp client session command in privileged EXEC mode. To set the value to the default, use the no form of this command.
show rtsp client session {history | active} [detailed]
no show rtsp client session {history | active} [detailed]
Syntax Description
history
|
Displays cumulative information about the session, packet statistics, and general call information such as call ID, session ID, individual RTSP stream URLs, packet statistics, and play duration.
|
active
|
If the keyword detailed is not specified, the command displays the session information and stream information for the stream that is currently active.
|
detailed
|
(Optional) If the keyword detailed is specified, the command displays the session information and stream information in detail for all streams that are associated with the session.
|
Defaults
Active (current) stream information is displayed.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use this command to display cumulative information about the session, packet statistics, and general call information such as call ID, session ID, and so on.
Note
Session refers to a session between the application and the RTSP client. Each call leg that is configured to use RTSP streaming has a session.
A call leg could play several prompts in a session; the "Play Time" refers to the play time associated with a stream or, in other words, a prompt; the cumulative play time is the sum total of all streams (or prompts) played out in a session.
The command output is a stream block that contains information about the stream (URL, packet statistics, current state of the stream, play duration, call ID, session ID, individual RTSP stream URLs, and packet statistics).
Examples
The following output is displayed when the show rtsp client session active command is used during an active session:
Router# show rtsp client session active
RTSP Session ID:0x8 Current Status:RTSP_STATUS_PLAYING
Active Request:RTSP_API_REQ_PLAY
Control Protocol:TCP Data Protocol:RTP
Total Packets Transmitted:0 (0 bytes)
Total Packets Received:708 (226560 bytes)
Cumulative Elapsed Play Time:00:00:28.296
Cumulative Elapsed Record Time:00:00:00.000
Session ID:0x8 State:ACTIVE
Local IP Address:1.13.79.45 Local Port 16660
Server IP Address:1.13.79.6 Server Port 11046
Stream URL:rtsp://rtsp-cisco.cisco.com:554/chinna.au/streamid=0
Packets Transmitted:0 (0 bytes)
Packets Received:708 (226560 bytes)
Elapsed Play Time:00:00:28.296
Elapsed Record Time:00:00:00.000
ReceiveDelay:85 LostPackets:0
The following output is displayed when the show rtsp client session history detailed command is used:
Router# show rtsp client session history detailed
Control Protocol:TCP Data Protocol:RTP
Total Packets Transmitted:0 (0 bytes)
Total Packets Received:2398 (767360 bytes)
Cumulative Elapsed Play Time:00:01:35.916
Cumulative Elapsed Record Time:00:00:00.000
Session ID:0x8 State:INACTIVE
Local IP Address:1.13.79.45 Local Port 16660
Server IP Address:1.13.79.6 Server Port 11046
Stream URL:rtsp://rtsp-cisco.cisco.com:554/chinna.au/streamid=0
Packets Transmitted:0 (0 bytes)
Packets Received:2398 (767360 bytes)
GapFillWithInterpolation:0
ReceiveDelay:85 LostPackets:0
EarlyPackets:2 LatePackets:12
Related Commands
Command
|
Description
|
rtsp client session history duration
|
Specifies the length of time the RTSP is kept during the session.
|
rtsp client session history records
|
Specifies the number of RTSP client session history records during the session.
|
show rudpv0 failures
To show SS7 Reliable User Datagram Protocol (RUDP) failure statistics, enter the show rudpv0 failures command in privileged EXEC mode.
show rudpv0 failures
Syntax Description
There are no keywords or arguments.
Defaults
There are no default behaviors or values.
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Examples
The following example shows the display of RUDP failures. The fields are self-explanatory.
**** RUDP Failure Stats ****
OptionNotSupportedFailures 0
EmptyBufferSendFailures 0
Related Commands
Command
|
Description
|
clear rudpv0 statistics
|
Resets the counters for the statistics generated by show rudpv0 failures to 0.
|
show rudpv0 statistics
|
Displays RUDP information about number of packets sent, received, and so forth. clear rudpv0 statistics resets the counters for these statistics to 0.
|
show rudpv0 statistics
To show SS7 Reliable User Datagram Protocol (RUDP) internal statistics, enter the
show rudpv0 statistics privileged EXEC command.
show rudpv0 statistics
Syntax Description
There are no keywords or arguments.
Defaults
There are no default behaviors or values.
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
Because the statistics counters are continually updated, the cumulative total may not be exactly equal to individual connection counters. After a connection is reset, previous statistics are lost, so the current connection statistics reflect only this instance of the RUDP connection—since the last reset.
Cumulative statistics reflect counts since the router was rebooted or since the last time the
clear rudpv0 statistics command was issued.
Examples
The following example shows the display of RUDP statistics and states for two connections. The fields are self-explanatory.
*** RUDP Internal Stats ****
Connection ID: 811641AC, Current State: OPEN
TotalPacketsReceived 4826
TotalDataPacketsReceived 1
Connection ID: 81163FD4, Current State: OPEN
TotalPacketsReceived 6755
TotalDataBytesSent 173690
TotalDataBytesReceived 56121
TotalDataPacketsSent 2695
TotalDataPacketsReceived 2265
Cumulative RudpV0 Statistics
TotalPacketsReceived 11581
TotalDataBytesSent 173690
TotalDataBytesReceived 56125
TotalDataPacketsSent 2695
TotalDataPacketsReceived 2266
Related Commands
Command
|
Description
|
clear rudpv0 statistics
|
Resets the counters for the statistics generated by show rudpv0 statistics to 0.
|
show rudpv0 failures
|
Displays RUDP information about failed connections and the reasons for them. clear rudpv0 statistics resets the counters for these statistics to 0.
|
show rudpv1
To display Reliable User Datagram Protocol (RUDP) information, use the show rudpv1 command in privileged EXEC mode.
show rudpv1 { failures | parameters | statistics }
Syntax Description
failures
|
RUDP failure statistics.
|
parameters
|
RUDP connection parameters.
|
statistics
|
RUDP internal statistics.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
Usage Guidelines
Because the statistics counters are continually updated, the cumulative total may not be exactly equal to individual connection counters. After a connection is reset, previous statistics are lost, so the current connection statistics reflect only this instance of the RUDP connection—since the last reset.
Cumulative statistics reflect counts since the router was rebooted or since the last time the clear rudpv1 statistics command was issued.
Examples
The following example shows sample output for show rudpv1 failures:
Router# show rudpv1 failures
**** RUDPV1 Failure Stats ****
CreateEventQueueFailure 0
OptionNotSupportedFailures 0
EmptyBufferSendFailures 0
The following example shows sample output for show rudpv1 parameters:
Router# show rudpv1 parameters
*** RUDPV1 Connection Parameters ***
Next Connection Id:61F72B6C, Remote conn id 126000
Null Seg Timeout 1000 1000
Trans State Timeout 2000 2000
Next Connection Id:61F72DAC, Remote conn id 126218
Null Seg Timeout 1000 1000
Trans State Timeout 2000 2000
The following example shows sample output for show rudpv1 statistics:
Router# show rudpv1 statistics
*** RUDPV1 Internal Stats ****
Connection ID:61F72B6C, Current State:OPEN
TotalDataBytesReceived 17808
TotalDataPacketsReceived 742
Connection ID:61F72DAC, Current State:OPEN
TotalDataPacketsReceived 0
Cumulative RudpV1 Statistics
TotalPacketsReceived 1047
TotalDataBytesReceived 18048
TotalDataPacketsReceived 752
Related Commands
Command
|
Description
|
clear rudpv1 statistics
|
Clears the RUDP statistics counters.
|
debug rudpv1
|
Displays debugging information for RUDP.
|
show settlement
To display the configuration for all settlement servers and see the specific provider and transactions, use the show settlement command in privileged EXEC mode. To reset to the default value, use the no form of this command.
show settlement [provider-number [transactions]]
no show settlement [provider-number [transactions]]
Syntax Description
provider-number
|
(Optional) Displays the attributes of a specific provider.
|
transactions
|
(Optional) Displays the transaction status of a specific provider.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Examples
The following example shows information about all settlement servers configured:
Address url = https://1.14.115.100:6556/
Encryption = all (default)
Max Concurrent Connections = 20 (default)
Connection Timeout = 3600 (s) (default)
Response Timeout = 1 (s) (default)
Retry Delay = 2 (s) (default)
Retry Limit = 1 (default)
Session Timeout = 86400 (s) (default)
Roaming = Disabled (default)
Number of Connections = 0
Number of Transactions = 7
The following example shows transaction and state information about a specific settlement server:
Router# show settlement 0 transactions
Transaction ID=8796304133625270342
state=OSPC_GET_DEST_SUCCESS, index=0
callingNumber=5710868, calledNumber=15125551212
Table 54 describes the significant fields shown in the display. The provider attributes not configured are not shown.
.
Table 54 show settlement Field Descriptions
Field
|
Description
|
type
|
Settlement provider type.
|
address url
|
URL address of the provider.
|
encryption
|
SSL encryption method.
|
max-connections
|
Maximum number of concurrent connections to provider.
|
connection-timeout
|
Connection timeout with provider (in seconds).
|
response-timeout
|
Response timeout with provider (in seconds).
|
retry-delay
|
Delay time between retries (in seconds).
|
retry-limit
|
Number of retries.
|
session-timeout
|
SSL session timeout (in seconds).
|
customer-id
|
Customer ID, assigned by provider.
|
device-id
|
Device ID, assigned by provider.
|
roaming
|
Roaming enabled.
|
signed-token
|
Indicates if the settlement token is signed by the server.
|
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time that a connection is maintained after a communication exchange is completed.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
show sgcp connection
To see all active SGCP connections on this router, use the show sgcp connection command in EXEC mode.
show sgcp connection [interface number]
Syntax Description
interface
|
(Optional) Specifies a DS1 interface.
|
number
|
(Optional) Specifies the T1 interface (controller) number. Valid values on the Cisco MC3810 multiservice concentrator are from 0 to 1.
|
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
Usage Guidelines
If you do not specify an interface, this command shows all the active SGCP connections on this host. If you specify an interface, this command shows only those active connections on the specified interface.
Examples
The following example shows the active connections on this router being displayed:
Router# show sgcp connection
Endpoint Call_ID(C) Conn_ID(I) (P)ort (M)ode (S)tate (E)vent[SIFL] (R)esult[EA]
1. ds1-0/1@r3810-5 C=1,1,2 I=0x1 P=16492,16476 M=3 S=4 E=3,0,0,3 R=0, 0
The following example shows the state of SGCP on the router being displayed:
Router# show sgcp connection
SGCP Admin State DOWN, Oper State DOWN
SGCP call-agent: 209.165.200.225 , SGCP graceful-shutdown enabled? FALSE
SGCP request timeout 40, SGCP request retries 10
Table 55 describes the significant fields shown in the display.
Table 55 show sgcp connection Field Descriptions
Field
|
Description
|
SGCP Admin State
|
The administrative and operational state of the SGCP daemon.
|
SGCP call-agent
|
The address of the call agent specified in the sgcp command.
|
SGCP graceful-shutdown enabled
|
The state of the sgcp graceful-shutdown command.
|
SGCP request timeout
|
The setting for the sgcp request timeout command.
|
SGCP request retries
|
The setting for the sgcp request retries command.
|
Related Commands
Command
|
Description
|
show sgcp endpoint
|
Displays SGCP endpoint information.
|
show sgcp statistics
|
Displays global statistics for the SGCP packet count, success, and failure counts.
|
show sgcp endpoint
To see SGCP endpoints eligible for SGCP management, use the show sgcp endpoint command in EXEC mode.
show sgcp endpoint [interface ds1 [ds0]]
Syntax Description
interface ds1
|
(Optional) Specifies the DS1 interface for which to display SGCP endpoint information. The valid range is from 1 to 1000.
|
ds0
|
(Optional) Specifies the DS0 interface for which to display SGCP endpoint information. The valid range is from 0 to 30.
|
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
Usage Guidelines
You can use this command to see SGCP endpoint information for the whole router, or you can see SGCP endpoint information for a specific DS1 interface and, optionally, a specific DS0. If you enter a nonexistent combination of a DS1 and DS0, the following error message appears: "No matching connection found."
Examples
The following command shows SGCP endpoint information being set for a matching connection between DS1 interface 1 and DS0 interface 10:
Router# show sgcp endpoint interface 1 10
Related Commands
Command
|
Description
|
show sgcp connection
|
Displays all the active connections on the host router.
|
show sgcp statistics
|
Displays global statistics for the SGCP packet count, success, and failure counts.
|
show sgcp statistics
To see global statistics for the SGCP packet count, success and failure counts, and other information, use the show sgcp statistics command in EXEC mode.
show sgcp statistics
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
Support for this command was extended to the Cisco MC3810 multiservice concentrator and the Cisco 3600 series routers (except for the Cisco 3620) in a private release that was not generally available.
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 universal access server only and was not generally available.
|
Examples
The following example shows SGCP packet statistics being displayed:
Router# show sgcp statistics
Unrecognized rx pkts 0, SGCP message parsing errors 0
Failed to send SGCP messages 0
CreateConn rx 1, successful 1, failed 0
DeleteConn rx 0, successful 0, failed 0
ModifyConn rx 0, successful 0, failed 0
DeleteConn tx 0, successful 0, failed 0
NotifyRequest rx 3, successful 3, failed 0
Notify tx 3, successful 3, failed 0
IP address based Call Agents statistics:
IP address 1.4.63.100, Total msg rx 5,
The following examples show how you can filter the command return for specific information:
Router# show sgcp statistics | begin Failed
Failed to send SGCP messages 0
CreateConn rx 0, successful 0, failed 0
DeleteConn rx 0, successful 0, failed 0
ModifyConn rx 0, successful 0, failed 0
DeleteConn tx 0, successful 0, failed 0
NotifyRequest rx 0, successful 0, failed 0
Notify tx 0, successful 0, failed 0
Router# show sgcp statistics | exclude ACK
Unrecognized rx pkts 0, SGCP message parsing errors 0
Failed to send SGCP messages 0
CreateConn rx 0, successful 0, failed 0
DeleteConn rx 0, successful 0, failed 0
ModifyConn rx 0, successful 0, failed 0
DeleteConn tx 0, successful 0, failed 0
NotifyRequest rx 0, successful 0, failed 0
Notify tx 0, successful 0, failed 0
Router# show sgcp statistics | include ACK
Related Commands
Command
|
Description
|
show sgcp connection
|
Display all the active connections on the host Cisco AS5300 universal access server.
|
show sgcp endpoint
|
Displays SGCP endpoint information.
|
show sip-ua
To display information and settings for the Session Initiation Protocol (SIP) User Agent (UA), use the show sip-ua command in privileged EXEC mode.
show sip-ua {retry | statistics | status | timers}
Syntax Description
retry
|
Displays SIP protocol retry counts.
|
statistics
|
Displays SIP UA response, traffic, and retry statistics.
|
status
|
Displays SIP UA listener status.
|
timers
|
Displays current settings for the SIP UA protocol timers.
|
Defaults
No default behaviors or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(3)T
|
The following changes were made:
• The statistics keyword was added.
• The statistics portion of the output from the status keyword was moved from the status keyword to the statistics keyword.
• The output from the timers keyword was changed to reflect the changes in the timers command.
|
Examples
The following example displays output for the show sip-ua retry command:
Router# show sip-ua retry
The following example displays output for the show sip-ua statistics command:
Router# show sip-ua statistics
SIP Response Statistics (Inbound/Outbound)
Forwarded 0/0, Queued 0/0,
OkCancel 0/0, OkOptions 0/0
Redirection (Inbound only):
MultipleChoice 0, MovedPermanently 0,
MovedTemporarily 0, SeeOther 0,
UseProxy 0, AlternateService 0
BadRequest 0/0, Unauthorized 0/0,
PaymentRequired 0/0, Forbidden 0/0,
NotFound 0/0, MethodNotAllowed 0/0,
NotAcceptable 0/0, ProxyAuthReqd 0/0,
ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
LengthRequired 0/0, ReqEntityTooLarge 0/0,
ReqURITooLarge 0/0, UnsupportedMediaType 0/0,
BadExtension 0/0, TempNotAvailable 0/0,
CallLegNonExistent 0/0, LoopDetected 0/0,
TooManyHops 0/0, AddrIncomplete 0/0,
Ambiguous 0/0, BusyHere 0/0
InternalError 0/0, NotImplemented 0/0,
BadGateway 0/0, ServiceUnavail 0/0,
GatewayTimeout 0/0, BadSipVer 0/0
BusyEverywhere 0/0, Decline 0/0,
NoExistAnywhere 0/0, NotAcceptable 0/0
SIP Total Traffic Statistics (Inbound/Outbound)
Invite 0/0, Ack 0/0, Bye 0/0,
Invite 0, Bye 0, Cancel 0, Response 0
The following example displays output for the show sip-ua status command:
Router# show sip-ua status
SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
The following example displays output for the show sip-ua timers command:
Router# show sip-ua timers
SIP UA Timer Values (millisecs)
trying 500, expires 180000, connect 500, disconnect 500
Related Commands
Command
|
Description
|
sip-ua
|
Enables the SIP user-agent configuration commands, with which you configure the user agent.
|
show ss7 mtp2 ccb
To display SS7 MTP 2 Channel Control Block (CCB) information, use the show ss7 mtp2 ccb command in privileged EXEC mode.
show ss7 mtp2 ccb [channel]
Syntax Description
channel
|
Specifies a channel from 0 through 3.
|
Defaults
The default is set when you first configure the MTP 2 variant. The link must be out of service in order to change the MTP 2 variant.
If you do not specify a channel, the command shows Channel Control Block information for channel 0.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The application and meaning of the output is dependent on the MTP 2 variant. For example, NTT and TTC only support emergency alignment.
Examples
The following example shows the display of MTP 2 CCB information:
Router# show ss7 mtp2 ccb 0
SS7 MTP2 Internal Channel Control Block Info for channel 0
Protocol version for channel 0 is Japan NTT Q.703 Version 1-1
ModuloSeqNumber = 128 (0x80 )
MaxSeqNumber = 127 (0x7F )
Unacked-MSUs (MaxInRTB) = 40 (0x28 )
MaxProvingAttempts = 5 (0x5 )
SUERM-threshold = 64 (0x40 )
SUERM-number-octets = 16 (0x10 )
SUERM-number-SUs = 256 (0x100 )
Tie-AERM-Emergency = 1 (0x1 )
Tin-AERM-Normal = 1 (0x1 )
MSU_FISU_Accepted_flag = FALSE
Abnormal_FIBR_flag = FALSE
local_processor_outage = FALSE
remote_processor_outage = FALSE
provingEmergencyFlag = FALSE
RemoteProvingEmergencyFlag = FALSE
further_proving_required = FALSE
ForceRetransmitFlag = FALSE
RetransmissionFlag = FALSE
show ss7 mtp2 state
To display internal SS7 Message Transfer Part level 2 (MTP 2) state machine information, use the show ss7 mtp2 state command in privileged EXEC mode.
show ss7 mtp2 state [channel]
Syntax Description
channel
|
Specifies a channel from 0 to 3.
|
Defaults
If you do not specify a channel, the command shows state machine information for channel 0.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Examples
The following examples show the display of MTP 2 state machine information for two different channels. Table 56 explains the fields.
Router# show ss7 mtp2 state 0
SS7 MTP2 states for channel 0
Protocol version for channel 0 is Japan NTT Q.703 Version 1-1
MTP2TXC_INSERVICE MTP2RC_IDLE
MTP2SUERM_IDLE MTP2AERM_IDLE
Congestion Backhaul = Abate
Remote Processor Outage = FALSE
Router# show ss7 mtp2 state 1
SS7 MTP2 states for channel 1
Protocol version for channel 1 is Japan NTT Q.703 Version 1-1
MTP2TXC_INSERVICE MTP2RC_IDLE
MTP2SUERM_IDLE MTP2AERM_IDLE
Congestion Backhaul = Abate
Remote Processor Outage = FALSE
Table 56 SS7 MTP 2 State Information Fields
State
|
Description
|
Possible Values
|
MTP2LSC
|
Indicates the overall status of the link.
|
OOS—The link is Out-of-Service.
INITIAL_ALIGNMENT—The link is in a transitional link alignment state.
ALIGNED_READY—The link is in a transitional link alignment state.
ALIGNED_NOT_READY—The link is in a transitional link alignment state.
INSERVICE—The link is in service.
PROCESSOR_OUTAGE—There is an outage in the local processor. This state implies that the link has been aligned.
POWER_OFF—It is possible you don't have the I/O memory set to at least 40 percent. There may not be enough memory for the SS7 MTP2 signaling.
|
MTP2IAC
|
Indicates the status of the initial alignment control state machine.
|
IDLE—The state machine is idle. It is not aligning the link.
NOT_ALIGNED—The state machine has begun the alignment process.
ALIGNED— The link has exchanged the alignment handshake with the remote device.
PROVING—The link alignment is being proven. This is a waiting period before the LSC state changes to INSERVICE.
|
MTP2TXC
|
Indicates the status of the transmission control state machine.
|
IDLE—The state machine is inactive.
INSERVICE—The state machine is the active transmitter.
|
MTP2RC
|
Indicates the status of the receive control state machine.
|
IDLE—The state machine is inactive.
INSERVICE—The state machine is the active receiver.
|
MTP2SUERM
|
Indicates the status of the signal unit error monitor (SUERM).
|
IDLE—The state machine is inactive.
MONITORING—The SUERM is active. SUERM uses a leaky-bucket algorithm to track link errors while the link is in service. If the number of link errors reaches the threshold, the link is taken out of service.
|
MTP2AERM
|
Indicates the status of the alignment error rate monitor state machine (AERM).
|
IDLE—The state machine is inactive.
MONITORING—Alignment error monitor is active. This is part of the alignment process.
|
MTP2CONGESTION
|
Indicates the status of the congestion control state machine.
|
IDLE—The state machine is inactive. No congestion is detected; normal traffic flow.
ACTIVE—Congestion has been declared. The Cisco 2600 series router is sending SIBs every T5, which indicates that the remote end should stop sending new MSUs until the local Cisco 2600 series router can catch up.
|
Congestion Backhaul
|
Indicates congestion status of the backhaul link between the Cisco SLT and the Media Gateway Controller.
|
Abate—The link between the Cisco 2600 series router and the Media Gateway Controller is not under congestion.
Onset—The link between the Cisco 2600 series router and the Media Gateway Controller is under congestion. and the Media Gateway Controller should stop sending new MSUs until the local Cisco 2600 series router can catch up.
|
Remote Processor Outage
|
Indicates the processor outage status of the remote.
|
TRUE indicates that the remote is in processor outage.
FALSE indicates that the remote has not declared processor outage.
|
show ss7 mtp2 stats
To display SS7 MTP 2 operational statistics, use the show ss7 mtp2 stats command in privileged EXEC mode.
show ss7 mtp2 stats [channel]
Syntax Description
channel
|
Specifies a channel from 0 through 3.
|
Defaults
If you do not specify a channel, the command shows status information for channel 0.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Examples
The following example shows operations and maintenance (OM) statistics for MTP 2 channel 0. Table 57 explains some of the fields.
Router# show ss7 mtp2 stats 0
SS7 MTP2 Statistics for channel 0
Protocol version for channel 0 is Japan NTT Q.703 Version 1-1
OMIACAlignAttemptCount = 0
OMIACAlignCompleteCount = 0
OMLSSU_XMIT_SIOSCount = 17
OMLSSU_XMIT_SIPOCount = 0
OMLSSU_RCV_InvalidCount = 0
OMRemote_Congestion_Cnt = 0
OMtimeNotINSV (secs) = 9550
OMMSUBytesTransmitted = 0
OMPDU_notAcceptedCount = 0
OMunreasonableFSN_rcvd = 0
OMunreasonableBSN_rcvd = 0
OMCongestionBackhaulCnt = 0
Table 57 SS7 OM Information Fields
Field
|
Description
|
OMIACAlignAttemptCount
OMIACAlignFailCount
OMIACAlignCompleteCount
|
Counts for Initial Alignment Control (IAC) attempts.
|
OMMSU_TO_XMIT_Count
|
This count is related to the results of the show ss7 sm stats command's PDU_pkts_recieve_count statistic. The number shown in OMMSU_TO_XMIT_Count is less than the PDU_pkts_recieve_count because OMMSU_TO_XMIT_Count shows the number of PDUs going out on the link, while the PDU_pkts_recieve_count includes PDUs that are internal to MTP2.
|
OMMSU_RCV_Count
|
Related to the results of the show ss7 sm stats command's packets_send_count.
|
OMLSSU_XMIT_Count
OMLSSU_XMIT_SINCount
OMLSSU_XMIT_SIECount
OMLSSU_XMIT_SIOCount
OMLSSU_XMIT_SIOSCount
OMLSSU_XMIT_SIPOCount
OMLSSU_XMIT_SIBCount
|
These counters represent the number of times that MTP 2 has posted the specific Link Status Signal Unit (LSSU) to MTP 1. They do not show the number of LSSUs actually sent over the link.
|
OMLSSU_RCV_Count
OMLSSU_RCV_SINCount
OMLSSU_RCV_SIECount
OMLSSU_RCV_SIOCount
OMLSSU_RCV_SIOSCount
OMLSSU_RCV_SIPOCount
OMLSSU_RCV_SIBCount
OMLSSU_RCV_InvalidCount
|
These counters represent the number of LSSUs received by MTP 2 from MTP 1. Because of MTP 1 filtering, this is not the same as the actual LSSUs sent over the link.
|
OMT1_TMO_Count
OMT2_TMO_Count
OMT3_TMO_Count
OMT4_TMO_Count
OMT5_TMO_Count
OMT6_TMO_Count
OMT7_TMO_Count
OMT8_TMO_Count
OMTA_TMO_Count
OMTF_TMO_Count
OMTO_TMO_Count
OMTA_TMO_Count
OMLostTimerCount
|
These fields show information about timers in use.
|
OMLostBackhaulMsgs
|
This count is related to the results of the show ss7 sm stats command's PDU_pkts_recieve_count statistic. The counter indicates how many messages received from the Media Gateway Controller have been lost because of a lack of resources in the Cisco 2600 series router. For example, if the Media Gateway Controller sends 100 MSUs and the Cisco 2600 series router only has 65 free buffers, 35 MSUs might be lost.
|
show ss7 mtp2 timer
To display durations of the SS7 MTP 2 state machine timers, use the show ss7 mtp2 timer command in privileged EXEC mode.
show ss7 mtp2 timer [channel]
Note
The eight timers whose status is displayed using the show ss7 mtp2 timer command are set on the Media Gateway Controller using MML commands. The timers are then downloaded from the controller to the Cisco SLT.
Syntax Description
channel
|
Specifies a channel from 0 through 3.
|
Defaults
If you do not specify a channel, the command shows status information for channel 0.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
MTP 2 uses eight different timers on each link. Throughout the link state transitions, multiple timers are active. An in-service MTP 2 link requires timers that are constantly started, stopped, and restarted. Use this command to display the configured timer durations.
Note
All MTP 2 configuration parameters are set at the Cisco SLT command line interface. The Media Gateway Controller parameter data files are no longer used to configure the Cisco SLT.
Examples
The following example shows how to display timer information for channel 0:
Router# show ss7 mtp2 timer 0
SS7 MTP2 Timers for channel 0 in milliseconds
Protocol version for channel 0 is Japan NTT Q.703 Version 1-1
T4 Emergency Proving = 3000
T7 excess ack delay = 2000
show ss7 mtp2 variant
To display information about the SS7 MTP 2 protocol variant, use the show ss7 mtp2 variant command in privileged EXEC mode.
show ss7 mtp2 variant [channel]
Syntax Description
channel
|
Specifies a channel from 0 through 3.
|
Defaults
If you do not specify a channel, the command shows protocol information for channel 0.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
Each country specifies its own variant of SS7, and the Cisco SLT supports several variants of the MTP 2 protocol. The selected variant can affect the MTP 2 statistics displayed by various commands. The Cisco SLT support the following variants:
•
Telcordia Technologies (formerly Bellcore)
•
ITU
•
NTT (Japan)
•
TTC (Japan Telecom)
Each channel can be configured to any one of the protocol variants. When you change from one variant to another, for example from Bellcore to NTT, the MTP 2 parameters default to those specified by NTT. You can then change the defaults as required.
Examples
The following example shows how to display protocol variant information for channel 1:
Router# show ss7 mtp2 variant 1
Protocol version for channel 1 is Bellcore GR-246-Core Issue 2, Dec 1997
show ss7 sm session
To display information about SS7 Session Manager session, use the show ss7 sm session command in privileged EXEC mode.
show ss7 sm session [session]
Syntax Description
session
|
Specifies a session, 0 or 1.
|
Defaults
If you do not specify a session, the command shows information for both sessions.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
If no sessions are configured, the message "No Session is configured" appears.
Examples
The following example shows how to display session information for both sessions. Table 58 explains the fields.
Router# show ss7 sm session
Session[0]: Remote Host 255.255.251.254:8060, Local Host 255.255.255.254:8060
Session[1]: Remote Host 255.255.251.255:8061, Local Host 255.255.255.254:8061
Table 58 Session Manager Session Information
Field
|
Description
|
Remote Host, Local Host
|
Shows the IP address and port number for the session.
|
retrans_t
|
Shows the retransmission timer value.
|
cumack_t
|
Shows the cumulative acknowledgment timer value.
|
m_cumack
|
Shows the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
m_outseq
|
Shows the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
m_rcvnum
|
Shows the maximum number of segments that the remote end can send before receiving an acknowledgment
|
Related Commands
Command
|
Description
|
ss7 session retrans_t
|
Sets the retransmission timer.
|
ss7 session m_rcvnum
|
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
|
ss7 session m_outseq
|
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
|
ss7 session m_cumack
|
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
|
ss7 session cumack_t
|
Sets the cumulative acknowledgment timer.
|
ss7 session
|
Establishes a session.
|
show ss7 sm set
To display information about the SS7 failover timer, use the show ss7 sm set command in privileged EXEC mode.
show ss7 sm set
Syntax Description
There are no arguments or keywords.
Defaults
There is no default.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Examples
The following example shows how to display failover timer information; the failover timer is set to the default of 3 seconds:
failover timer = 3 seconds
Related Commands
Command
|
Description
|
ss7 set failover timer
|
Specifies the amount of time that the Session Manager waits for the session to recover before declaring the session inactive.
|
ss7 session
|
Establishes a session.
|
show ss7 sm stats
To display SS7 Session Manager session statistics, use the show ss7 sm stats command in privileged EXEC mode.
show ss7 sm stats
Syntax Description
There are no arguments or keywords for this command.
Defaults
The command shows information for both sessions.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR
|
This command was introduced.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
If no sessions are configured, the message "No Session is configured" appears.
Examples
The following example shows how to display SS7 Session Manager statistics. The fields are self-explanatory and show information about the session state, protocol data units (PDUs) packets sent and received, and SS7 Reliable User Datagram Protocol (RUDP) performance:
Router# show ss7 sm stats
-------------------- Session Manager --------------------
Session Manager state = SESSION SET STATE-ACTIVE
Session Manager Up count = 1
Session Manager Down count = 0
lost control packet count = 0
failover timer expire count = 0
invalid_connection_id_count = 0
Session[0] statistics SM SESSION STATE-STANDBY:
Total Pkts receive count = 1
Active Pkts receive count = 0
Standby Pkts receive count = 1
PDU Pkts receive count = 0
Unknown Pkts receive count = 0
-Pkts window full count = 0
-Pkts resource unavail count = 0
-Pkts enqueue fail count = 0
RUDP Connection Not Open = 0
RUDP Invalid Conn Handle = 0
NonActive Receive count = 0
Session[1] statistics SM SESSION STATE-ACTIVE:
Total Pkts receive count = 2440
Active Pkts receive count = 1
Standby Pkts receive count = 0
PDU Pkts receive count = 2439
Unknown Pkts receive count = 0
-Pkts window full count = 0
-Pkts resource unavail count = 0
-Pkts enqueue fail count = 0
RUDP Connection Not Open = 0
RUDP Invalid Conn Handle = 0
NonActive Receive count = 0
Related Commands
Command
|
Description
|
clear ss7 sm-stats
|
Clears the counters that track Session Manager statistics for the show ss7 sm stats command.
|
ss7 session
|
Establishes a session.
|
show translation-rule
To display the contents of the rules that have been configured for a specific translation name, use the show translation-rule command in privileged EXEC mode.
show translation-rule [name-tag]
Syntax Description
name-tag
|
(Optional) The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is from 1 through 2,147,483,647.
|
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for Voice over IP on the Cisco AS5300 universal access server.
|
12.0(7)XK
|
This command was first supported for the following voice technologies on the following platforms:
• Voice over IP (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over Frame Relay (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over ATM (Cisco 3600 series, Cisco MC3810 multiservice concentrator)
|
12.1(1)T
|
This command was first supported on the T train for the following voice technology on the following platforms:
• Voice over IP (1750, Cisco 2600 series, Cisco 3600 series, Cisco AS5300 universal access server, Cisco 7200 series, and Cisco 7500 series)
|
12.1(2)T
|
This command was first supported on the T train for the following voice technologies on the following platforms:
• Voice over IP (Cisco MC3810 multiservice concentrator)
• Voice over Frame Relay (Cisco 2600 series, Cisco 3600 series, Cisco MC3810 multiservice concentrator)
• Voice over ATM (Cisco 3600 series, Cisco MC3810 multiservice concentrator)
|
Usage Guidelines
This command gives detailed information about the configured rules under this rule name. If the name tag is not entered, a complete display of all the configured rules will be shown.
Examples
The following example shows output for the show translation-rule command:
Router# show translation-rule
Translation rule address:0x61AB94F8
Translation rule in_used 1
**** Xrule rule table *******
**** Xrule rule table *******
Translation rule address:0x61C2E6D4
Translation rule in_used 1
**** Xrule rule table *******
Table 59 describes the significant fields shown in the display.
Table 59
Translation rule address
|
The translation rule address in hex.
|
Tag name
|
The translation rule tag name.
|
Translation rule in_used
|
The translation rule in which the tag is used.
|
**** Xrule rule table *******
|
Specifies the beginning of the display for a specific rule.
|
Rule:x
|
The number of the rule.
|
in_used state:
|
The input-searched-pattern.
|
Match pattern:
|
The match pattern of the rule.
|
Sub pattern:
|
The substituted pattern.
|
Match type:
|
The match type.
|
Sub type:
|
The substituted pattern match type.
|
show translation-rule Field Descriptions
Related Commands
Command
|
Description
|
numbering-type
|
Specifies number type for the VoIP or POTS dial peer.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
test translation-rule
|
Tests the execution of the translation rules on a specific name-tag.
|
translate
|
Applies a translation rule to a calling party number or a called party number for incoming calls.
|
translate-outgoing
|
Applies a translation rule to a calling party number or a called party number for outgoing calls.
|
translation-rule
|
Creates a translation name and enters translation-rule configuration mode.
|
voip-incoming translation-rule
|
Captures calls that originate from H.323-compatible clients.
|
show vfc
To see the entries in the host-name-and-address cache, use the show vfc command in privileged EXEC mode.
show vfc slot-number [technology]
Syntax Description
slot-number
|
VFC slot number.
|
technology
|
(Optional) Displays the technology type of the VFC.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3 NA
|
This command was introduced on the Cisco AS5300 universal access server.
|
12.0(2)XH
|
The technology keyword was added.
|
Examples
The following example shows that the card in slot 1 is a C549 DSPM:
Router# show vfc 1 technology
Technology in VFC slot 1 is C549
Related Commands
Command
|
Description
|
voice-card
|
Configures a voice card and enters voice-card configuration mode.
|
show vfc cap-list
To show the current list of files on the capability list for this voice feature card (VFC), use the show vfc cap-list command in user EXEC mode.
show vfc slot cap-list
Syntax Description
slot
|
Identifies the slot where the VFC is installed. Valid entries are from 0 to 2.
|
Defaults
No default behavior or values.
Command Modes
User EXEC
Command History
Release
|
Modification
|
11.3 NA
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
To identify the specific VFC, enter the number of the slot on the chassis where the VFC resides using the slot argument.
Examples
The following is sample output from the show vfc cap-list command:
Router# show vfc 1 cap-list
Capability List for VFC in slot 1:
The first line in this output is a general description, stating that this is the capability list for the VFC residing in slot 1. Below this is a numbered list, each line of which identifies one currently installed in-service file.
Related Commands
Command
|
Description
|
show vfc default-file
|
Displays the default files included in the default file list for this VFC.
|
show vfc directory
|
Displays the list of all files residing on this VFC.
|
show vfc version
|
Displays the version of the software residing on this VFC.
|
show vfc default-file
To show the default files included in the default file list for a voice feature card (VFC), use the show vfc default-file command in user EXEC mode.
show vfc slot default-file
Syntax Description
slot
|
Identifies the slot where the VFC is installed. Valid entries are from 0 to 2.
|
Defaults
No default behavior or values.
Command Modes
User EXEC
Command History
Release
|
Modification
|
11.3 NA
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use the show vfc default-file user EXEC command to display a list of all default files for a particular voice feature card. To identify the specific VFC, enter the number of the slot on the chassis where the VFC resides using the slot argument.
Examples
The following is sample output from the show vfc default-file command:
Router# show vfc 1 default-file
Default List for VFC in slot 1:
The first line in this output is a general description, stating that this is the default list for the VFC residing in slot 1. Below this is a numbered list, each line of which identifies one default file.
Related Commands
Command
|
Description
|
show vfc cap-list
|
Displays the current list of files on the capability list for this VFC.
|
show vfc directory
|
Displays the list of all files residing on this VFC.
|
show vfc version
|
Displays the version of the software residing on this VFC.
|
show vfc directory
To show the list of all files residing on a voice feature card (VFC), use the show vfc directory command in user EXEC mode.
show vfc slot directory
Syntax Description
slot
|
Identifies the slot where the VFC is installed. Valid entries are from 0 to 2.
|
Defaults
No default behavior or values.
Command Modes
User EXEC
Command History
Release
|
Modification
|
11.3 NA
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use the show vfc directory user EXEC command to display a list of all of the files currently stored in Flash memory for a particular VFC. To identify the specific VFC, enter the number of the slot on the chassis where the VFC resides using the slot argument.
Examples
The following is sample output from the show vfc directory command:
Router# show vfc 1 directory
Files in slot 1 VFC flash:
1 . vcw-vfc-mz.gsm.VCW 292628
2 . btl-vfc-l.0.13.0.bin 4174
3 . cor-vfc-l.0.1.bin 54560
4 . jbc-vfc-l.0.13.0.bin 16760
5 . fax-vfc-l.0.1.bin 64290
6 . bas-vfc-l.0.1.bin 54452
7 . cdc-g711-l.0.1.bin 190
8 . cdc-g729-l.0.1.bin 21002
9 . cdc-g726-l.0.1.bin 190
10. cdc-g728-l.0.1.bin 22270
11. cdc-gsmfr-l.0.1.bin 190
Table 60 describes the significant fields shown in the display.
Table 60 show vfc directory Field Descriptions
Field
|
Description
|
File Name
|
Name of the file stored in Flash memory.
|
Size (Bytes)
|
Size of the file in bytes.
|
Related Commands
Command
|
Description
|
show vfc cap-list
|
Displays the current list of files on the capability list for this VFC.
|
show vfc default-file
|
Displays the default files included in the default file list for this VFC.
|
show vfc version
|
Displays the version of the software residing on this VFC.
|
show vfc version
To show the version of the software residing on a voice feature card (VFC), use the show vfc version command in user EXEC mode.
show vfc slot version {dspware | vcware}
Syntax Description
slot
|
Identifies the slot where the VFC is installed. Valid values are 0, 1, and 2.
|
dspware
|
Defines which DSPWare software to display.
|
vcware
|
Defines which VCWare software to display.
|
Defaults
No default behavior or values.
Command Modes
User EXEC
Command History
Release
|
Modification
|
11.3 NA
|
This command was introduced on the Cisco AS5300 universal access server.
|
Usage Guidelines
Use the show vfc version user EXEC command to display the version of the software (running on either DSP or VFC) currently installed in Flash memory on the VFC.
Examples
The following is sample output from the show vfc version command:
Router# show vfc 0 version dspware
Version of Dspware in VFC slot 0 is 0.10
The output from this command is a simple declarative sentence stating the version number for the selected type of software (in this example, DSPWare) for the VFC residing in the selected slot number (in this example, slot 0).
Related Commands
Command
|
Description
|
show vfc cap-list
|
Displays the current list of files on the capability list for this VFC.
|
show vfc default-file
|
Displays the default files included in the default file list for this VFC.
|
show vfc directory
|
Displays the list of all files residing on this VFC.
|
show video call summary
To display summary information about video calls and the current status of the Video Call Manager (ViCM), use the show video call summary command in privileged EXEC mode.
show video call summary
Syntax Description
There are no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(5)XK
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)T
|
The command introduced in Cisco IOS Release 12.0(5)XK was integrated into Cisco IOS Release 12.0(7)T.
|
Usage Guidelines
Use this command to look quickly at the status of current calls. In Cisco IOS Releases 12.0(5)XK and 12.0(7)T, there can be only one video call in progress.
Examples
On a Cisco MC3810 multiservice concentrator, the following example displays information about the ViCM when no call is in progress on the serial interface that connects to the local video codec:
Router# show video call summary
Serial0:ViCM = Idle, Codec Ready
When a call is starting, the output looks like this:
Router# show video call summary
Serial0:ViCM = Call Connected
When a call is disconnecting, the output looks like this:
Router# show video call summary
Related Commands
Command
|
Description
|
show call history video record
|
Displays information about video calls.
|
show voice busyout
To display information about the voice busyout state, use the show voice busyout command in privileged EXEC mode.
show voice busyout
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
The show voice busyout command lists the following information:
•
Interfaces that are being monitored for busyout events
•
Voice ports currently in the busyout state and the reasons
Examples
The following example displays the busyout information:
Router# show voice busyout
If following network interfaces are down, voice port will be put into busyout state
ATM0
Serial0
The following voice ports are in busyout state
1/1 is forced into busyout state
1/2 is in busyout state caused by network interfaces
1/3 is in busyout state caused by ATM0
1/4 is in busyout state caused by network interfaces
1/5 is in busyout state caused by Serial0
Related Commands
Command
|
Description
|
busyout forced
|
Forces a voice port into the busyout state.
|
busyout monitor
|
Places a voice port in the busyout monitor state.
|
busyout seize
|
Changes the busyout seize procedure from a voice port.
|
voice-port busyout
|
Places all voice ports associated with a serial or ATM interface in a busyout state.
|
show voice call
To show the call status for voice ports on the Cisco router or concentrator, use the show voice call EXEC command.
Cisco 2600 and 3600 series with Analog Voice Ports
show voice call [slot/subunit/port | summary]
Cisco 2600 and 3600 Series with Digital Voice Ports (with T1 Packet Voice Trunk Network Modules)
show voice call [slot/port:ds0-group | summary]
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
show voice call [slot/port | summary]
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
show voice call [slot:ds0-group | summary]
Syntax Description
For the Cisco 2600 and 3600 Series with Analog Voice Ports:
slot/subunit/port
|
(Optional) Displays information for the analog voice port you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.)
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco 2600 and 3600 Series with Digital Voice Ports:
slot/port:ds0-group
|
(Optional) Displays information for the digital voice port you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. (One VWIC fits in an NM.)
• ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
(Optional) Displays information for the analog voice port you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810 multiservice concentrator.
• port specifies an analog voice port number. Valid entries are from 1 to 6.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
(Optional) Displays information for the digital voice port you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
Command Modes
EXEC
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced for the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
The show voice call command applies to Voice over Frame Relay, Voice over ATM, and Voice over IP.
This command shows call-processing and protocol state-machine information for a voice port, if it is available. It also shows information on the DSP channel associated with the voice port, if it is available. All real-time information in the DSP channel, such as jitter and buffer overrun for example, is queried to the DSP channel, and asynchronous responses are returned to the host side.
If no call is active on a voice port, the show voice call summary command displays only the VPM (shutdown) state. If a call is active on a voice port, the VTSPS state is shown. For an on-net call or a local call without local-bypass (not cross-connected), the CODEC and VAD fields are displayed. For an off-net call or a local call with local-bypass, the CODEC and VAD fields are not displayed.
CODEC and VAD are not displayed by the show voice call command because this information is in the summary display.
The show voice call command provides the status at these levels of the call handling module:
•
Tandem switch
•
End-to-end call manager
•
Call processing state machine
•
Protocol state machine
Examples
The following is a sample display from the show voice call summary command for voice ports on a Cisco MC3810 multiservice concentrator, showing two local calls connected without local bypass:
PORT CODEC VAD VTSP STATE VPM STATE
======= ======== === ===================== ========================
0:18.19 g729ar8 n S_CONNECT FXOLS_OFFHOOK
1/6 g729ar8 n S_CONNECT FXOLS_CONNECT
The following is a sample display from the show voice call summary command for voice ports on a Cisco MC3810 multiservice concentrator, showing two local calls connected with local bypass:
PORT CODEC VAD VTSP STATE VPM STATE
======= ======== === ===================== ========================
0:18.19 S_CONNECT FXOLS_OFFHOOK
1/6 S_CONNECT FXOLS_CONNECT
The following is a sample display from the show voice call command for analog voice ports on a Cisco MC3810 multiservice concentrator:
1/1 vpm level 1 state = FXSLS_ONHOOK
1/2 vpm level 1 state = FXSLS_ONHOOK
1/4 vtsp level 0 state = S_CONNECT
vpm level 1 state = S_TRUNKED
1/5 vpm level 1 state = EM_ONHOOK
1/6 vpm level 1 state = EM_ONHOOK
sys252#show voice call 1/4
1/4 vtsp level 0 state = S_CONNECT
vpm level 1 state = S_TRUNKED
router# ***DSP VOICE VP_DELAY STATISTICS***
Clk Offset(ms): 1445779863, Rx Delay Est(ms): 95
Rx Delay Lo Water Mark(ms): 95, Rx Delay Hi Water Mark(ms): 125
***DSP VOICE VP_ERROR STATISTICS***
Predict Conceal(ms): 10, Interpolate Conceal(ms): 0
Silence Conceal(ms): 0, Retroact Mem Update(ms): 0
Buf Overflow Discard(ms): 20, Talkspurt Endpoint Detect Err: 0
***DSP VOICE RX STATISTICS***
Rx Vox/Fax Pkts: 537, Rx Signal Pkts: 0, Rx Comfort Pkts: 0
Rx Dur(ms): 50304730, Rx Vox Dur(ms): 16090, Rx Fax Dur(ms): 0
Rx Non-seq Pkts: 0, Rx Bad Hdr Pkts: 0
Rx Early Pkts: 0, Rx Late Pkts: 0
***DSP VOICE TX STATISTICS***
Tx Vox/Fax Pkts: 567, Tx Sig Pkts: 0, Tx Comfort Pkts: 0
Tx Dur(ms): 50304730, Tx Vox Dur(ms): 17010, Tx Fax Dur(ms): 0
***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0
TDM Bus Levels(dBm0): Rx -70.3 from PBX/Phone, Tx -68.0 to PBX/Phone
TDM ACOM Levels(dBm0): +2.0, TDM ERL Level(dBm0): +5.6
TDM Bgd Levels(dBm0): -71.4, with activity being voice
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays the configuration for all VoIP and POTS dial peers configured on the router.
|
show voice dsp
|
Displays the current status of all DSP voice channels.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show voice dsp
To show the current status of all digital signal processor (DSP) voice channels, use the show voice dsp command in privileged EXEC mode.
show voice dsp
Syntax Description
This command has no arguments or keywords.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(7)XK
|
This command was first supported on the Cisco 2600 and 3600 series routers, and the display format was modified.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
Use this command when abnormal behavior in the DSP voice channels occurs.
Examples
The following is sample output from the show voice dsp command on a Cisco MC3810 multiservice concentrator:
DSP# 0, channel# 0 G729A BUSY
DSP# 0, channel# 1 G729A BUSY
DSP# 1, channel# 2 FAX IDLE
DSP# 1, channel# 3 FAX IDLE
DSP# 2, channel# 4 NONE BAD
DSP# 2, channel# 5 NONE BAD
DSP# 3, channel# 6 NONE BAD
DSP# 3, channel# 7 NONE BAD
DSP# 4, channel# 8 NONE BAD
DSP# 4, channel# 9 NONE BAD
DSP# 5, channel# 10 NONE BAD
DSP# 5, channel# 11 NONE BAD
Table 61 describes the significant fields shown in the display.
Table 61 show voice dsp Field Descriptions
Field
|
Description
|
DSP
|
Number of the DSP.
|
Channel
|
Number of the channel and its status.
|
The following is sample output from the show voice dsp command on a Cisco 1750 router:
DSP#0: state IN SERVICE, 2 channels allocated
channel#0: voice port 1/0, codec G711 ulaw, state UP
channel#1: voice port 1/1, codec G711 ulaw, state UP
DSP#1: state IN SERVICE, 2 channels allocated
channel#0: voice port 2/0, codec G711 ulaw, state UP
channel#1: voice port 2/1, codec G711 ulaw, state UP
DSP#2: state RESET, 0 channels allocated
Table 62 describes the significant fields shown in the display.
Table 62 show voice dsp Field Descriptions
Field
|
Description
|
DSP
|
Number of the DSP.
|
Channel
|
Number of the channel and its status.
|
Related Commands
Command
|
Description
|
clear counters
|
Clears all the current interface counters from the interface.
|
show dial-peer voice
|
Displays configuration information for dial peers.
|
show voice call
|
Displays the call status for all voice ports.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show voice permanent-call
To display information about the permanent calls on a voice interface, use the show voice permanent-call command in user EXEC or privileged EXEC mode.
show voice permanent-call [voice-port] [summary]
Syntax Description
voice-port
|
(Optional) Slot number or slot/port number of the voice interface for which you wish to display permanent call information.
|
summary
|
(Optional) Displays summary information about VoFR and VoATM ports used for permanent connections.
|
Defaults
No default behavior or values.
Command Modes
User EXEC
Privileged EXEC
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator.
|
12.0(4)T
|
The command introduced in Cisco IOS Release 12.0(3)XG was integrated into Cisco IOS Release 12.0(4)T.
|
Usage Guidelines
This command is available only on the Cisco MC3810 multiservice concentrator.
When no parameters are specified with this command, the output displays information for all ports containing permanent calls. When a specific interface is specified, information is displayed about the permanent calls for that interface only.
Examples
The following is sample output for the show voice permanent-call command:
Router# show voice permanent-call 1/1
1/1 state=connect coding=G729A payload size=30 vad=off
ec=8 (ms), cng=off fax=on digit_relay=on Seq num = off, VOFR Serial0,dlci = 550,cid = 6
TX INFO :slow-mode seq#= 25, sig pkt cnt= 19646, last-ABCD=1101
hardware-state ACTIVE signal type is CEPT/MELCAS
voice-gate CLOSED,network-path OPEN MASTER
1101 1101 1101 1101 1101 1101 1101 1101 1101 1101
1101 1101 1101 1101 1101 1101 1101 1101 1101 1101
1101 1101 1101 1101 1101 1101 1101 1101 1101 1101
RX INFO :slow-mode, sig pkt cnt= 19648, under-run = 0, over-run = 0
missing = 0, out of seq = 0, very late = 0
playout depth = 0 (ms), refill count = 1
prev-seq#= 25, last-ABCD=1101, slave standby timeout 25000 (ms)
max inter-arrival time 0 (ms), current timer 384 (ms)
max timeout timer 5016 (ms), restart timeout is 0 (ms)
signaling packet fast-mode inter-arrival times (ms)
16 24 16 24 16 24 16 24 16 24 16 24 16 24 16 24
16 24 16 24 16 24 16 24 0 0 0 0 0 0 0 0
signaling playout history
1101 1101 1101 1101 1101 1101 1101 1101 1101 1101
1101 1101 1101 1101 1101 1101 1101 1101 1101 1101
1101 1101 1101 1101 1101 1101 1101 1101 1101 1101
The following is sample output for the show voice permanent-call summary command:
Router# show voice permanent-call summary
1/1 state= connect, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 880,cid = 6
1/2 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 102
1/3 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 103
1/4 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 104
1/5 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 105
1/6 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 106
1/7 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 107
1/8 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 108
1/9 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 109
1/10 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 110
1/11 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 111
1/12 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 112
1/13 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 113
1/14 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 114
1/15 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 115
1/17 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 117
1/18 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 118
1/19 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 119
1/20 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 120
1/21 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 121
1/22 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 122
1/23 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 123
1/24 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 124
1/25 state= frf11, coding=G729A, payload size=30, vad=off, ec=64, cng=off, fax=on
digit_relay=off, VOFR Serial0:1,dlci = 990,cid = 125
Table 63 describes the fields shown in these displays.
Table 63 show voice permanent-call Field Descriptions
Field
|
Description
|
state
|
Current status of the call on this voice port.
|
coding
|
Codec type used for this call.
|
payload size
|
Size in bytes of the voice payload.
|
vad
|
Indicates whether voice activity detection is turned on or off.
|
ec
|
Echo canceler length, in milliseconds.
|
cng
|
Indicates whether or not comfort noise generation is used.
|
fax
|
Indicates if fax-relay is enabled.
|
digit_relay
|
Indicates if FRF.11 Annex A DTMF digit-relay is enabled.
|
Seq num
|
Indicates whether sequence numbers are turned on or off.
|
VOFR
|
Interface used for this call.
|
dlci
|
DLCI for this call.
|
cid
|
DLCI subchannel for this call.
|
TX INFO:slow-mode
|
Indicates that FRF.11 Annex B packets are being sent at the slow rate defined by the signal timing keepalive period.
|
TX INFO:seq#
|
Sequence number of the last packet sent.
|
TX INFO:sig pkt cnt
|
Number of signaling packets sent by this dial peer.
|
TX INFO:last-ABCD
|
Last ABCD signaling state sent by this dial peer to the network.
|
hardware-state
|
Indicates the on-hook/off-hook state of the call when the signaling protocol in use is a supported protocol. Not valid when the signal type is "transparent."
|
signal type
|
Indicates the type of call-control signaling used by this dial peer.
|
voice-gate
|
Indicates whether voice packets are being sent (OPEN) or not sent (CLOSED).
|
network-path
|
Indicates if any type of packet is being sent (OPEN) or not sent (CLOSED) to the network. This field will indicate CLOSED only if the port is configured as a slave using the connection trunk answer-mode command.
|
RX INFO:slow-mode
|
Indicates that FRF.11 Annex B packets are being received at the slow rate. Successive packets have the same sequence number.
|
RX INFO:sig pkt cnt
|
Number of slow-mode signaling packets received by this dial peer.
|
RX INFO:under-run
|
Valid for fast-mode only. Counts the number of times the signaling playout buffer became empty during FRF.11 Annex B fast-mode. In this mode, signaling packets are expected to be received every 20 milliseconds.
|
RX INFO:over-run
|
Valid for fast-mode only. Counts the number of times the signaling playout buffer became full during FRF.11 Annex B fast-mode. In this mode, signaling packets are expected to be received every 20 milliseconds.
|
RX INFO:missing
|
Indicates the number of FRF.11 Annex B packets that were counted as missing based on checking Annex B sequence numbers.
|
RX INFO:out of seq
|
Number of FRF.11 Annex B packets that were counted as received in the wrong order based on checking Annex B sequence numbers.
|
RX INFO:very late
|
Number of FRF.11 Annex B packets that were received with a sequence number significantly different from the expected sequence number.
|
RX INFO:playout depth
|
Valid for fast-mode only. Shows the current FRF.11 Annex B signaling buffer playout depth in milliseconds.
|
RX INFO:refill count
|
Indicates the number of times the FRF.11 Annex B signaling playout buffer was refilled as a result of a slow-mode to fast-mode transition.
|
RX INFO:prev-seq#
|
Sequence number of the last FRF.11 Annex B signaling packet received.
|
RX INFO:last-ABCD
|
Last ABCD signaling bit pattern sent to the attached PBX (telephone network side). In the out-of-service condition, this will show the OOS pattern being sent to the PBX.
|
RX INFO:slave standby timeout
|
Value configured using the signal timing oos standby command for the applicable voice class permanent entry.
|
max inter-arrival time
|
Maximum interval between the arrival of fast-mode FRF.11 Annex B packets since the last time this parameter was displayed.
|
current timer
|
Time, in milliseconds, since the last signaling packet was received.
|
max timeout timer
|
Maximum value of the "current timer" parameter since the last time it was displayed.
|
restart timeout
|
Connection restart timeout value.
|
signaling packet fast-mode inter-arrival time
|
Shows the last several values of the fast-mode FRF.11 Annex B signaling packet inter-arrival time.
|
signaling playout history
|
Shows recent ABCD signaling bits received from the data network.
|
Related Commands
Command
|
Description
|
show frame-relay fragment
|
Displays Frame Relay fragmentation details.
|
show frame-relay pvc
|
Displays statistics about PVCs for Frame Relay interfaces.
|
show frame-relay vofr
|
Displays details about FRF.11 subchannels being used on Voice over Frame Relay DLCIs.
|
show voice port
To display configuration information about a specific voice port, use the show voice port command in EXEC command.
Cisco 1750 Router
show voice port slot-number/port
Cisco 2600 and 3600 Series Router with Analog Voice Ports
show voice port [slot/subunit/port | summary]
2600 and 3600 Series Router with Digital Voice Ports (with T1 Packet Voice Trunk Network Modules)
show voice port [slot/port:ds0-group | summary]
Cisco MC3810 Multiservice Concentrator with Analog Voice Ports
show voice port [slot/port | summary]
Cisco MC3810 Multiservice Concentrator with Digital Voice Ports
show voice port [slot:ds0-group | summary]
Cisco AS5300 Universal Access Server
show voice port controller number:D
Cisco AS5800 Universal Access Server
show voice port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
Cisco 7200 Series Router
show voice port {slot/port:ds0-group-no} | {slot-number/subunit-number/port}
Syntax Description
For the Cisco 1750 Router:
slot-number
|
Slot number in the router where the voice interface card (VIC) is installed. Valid entries are from 0 to 2, depending on the slot where it has been installed.
|
port
|
Indicates the voice port. Valid entries are 0 or 1.
|
For the Cisco 2600 and 3600 Series Router with Analog Voice Ports:
slot/subunit/port
|
(Optional) Displays information for the analog voice port you specify with the slot/subunit/port designation.
• slot specifies a router slot in which a voice network module (VNM) is installed. Valid entries are router slot numbers for the particular platform.
• subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.)
• port specifies an analog voice port number. Valid entries are 0 and 1.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco 2600 and 3600 Series Router with Digital Voice Ports:
slot/port:ds0-group
|
(Optional) Displays information for the digital voice port you specify with the slot/port:ds0-group designation.
• slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.
• port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. (One VWIC fits in an NM.)
• ds0-group specifies a T1 or E1 logical port number. Valid entries are from 0 to 23 for T1 and from 0 to 30 for E1.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco MC3810 Multiservice Concentrator with Analog Voice Ports:
slot/port
|
(Optional) Displays information for the analog voice port you specify with the slot/port designation.
• slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810 multiservice concentrator.
• port specifies an analog voice port number. Valid entries are from 1 to 6.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco MC3810 Multiservice Concentrator with Digital Voice Ports:
slot:ds0-group
|
(Optional) Displays information for the digital voice port you specify with the slot:ds0-group designation.
• slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1).
• ds0-group specifies a T1 or E1 logical voice port number. Valid entries are from 0 to 23 for T1 and from 0 to 30 for E1.
|
summary
|
(Optional) Displays a summary of all voice ports.
|
For the Cisco AS5300 Access Server :
controller number
|
Specifies the T1 or E1 controller.
|
:D
|
Indicates the D channel associated with ISDN PRI.
|
For the Cisco AS5800 Universal Access Server:
shelf/slot/port
|
Specifies the T1 or E1 controller on the T1 card. Valid entries for the shelf argument are from 0 to 9999. Valid entries for the slot argument are from 0 to 11. Valid entries for the port argument are from 0 to 11.
|
shelf/slot/parent:port
|
Specifies the T1 controller on the T3 card. Valid entries for the shelf argument are from 0 to 9999. Valid entries for the slot argument are from 0 to 11. Valid entries for the port argument is 1 to 28. The value for the parent argument is always 0.
|
:D
|
Indicates the D channel associated with ISDN PRI.
|
For the Cisco 7200 Series Router:
slot
|
The router location where the voice port adapter is installed. Valid entries are from 0 to 3.
|
port
|
Indicates the voice interface card location. Valid entries are 0 and 1.
|
dso-group-no
|
Indicates the defines DS0 group number. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
|
slot-number
|
Indicates the slot number in the Cisco router where the voice interface card is installed. Valid entries are from 0 to 3, depending on the slot where it has been installed.
|
subunit-number
|
Indicates the subunit on the voice interface card where the voice port is located. Valid entries are 0 and 1.
|
port
|
Indicates the voice port number. Valid entries are 0 and 1.
|
Defaults
No default behavior or values.
Command Modes
EXEC
Command History
Release
|
Modification
|
11.3(1) T
|
This command was introduced on the Cisco 3600 series router.
|
11.3(1)MA
|
Port-specific values for the Cisco MC3810 multiservice concentrator were added.
|
12.0(3)T
|
Port-specific values for the Cisco MC3810 multiservice concentrator were added.
|
12.0(5)XK
|
The ds0-group argument was added for the Cisco 2600 and Cisco 3600 series routers.
|
12.0(5)XE
|
Additional syntax was created for digital voice to allow specification of the DS0 group. This command applies to VoIP on the Cisco 7200 series.
|
12.0(7)T
|
The additions from Cisco IOS Release 12.0(5)XE were integrated into Cisco IOS Release 12.0(7)T.
|
12.0(7)XK
|
The summary keyword was added for the Cisco 2600 and 3600 series routers. The ds0-group argument was added for the Cisco MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
Usage Guidelines
Use the show voice port EXEC command to display configuration and voice-interface-card-specific information about a specific port.
This command applies to Voice over IP, Voice over Frame Relay, and Voice over ATM.
The ds0-group command automatically creates a logical voice port that is numbered as follows on Cisco 7200 series routers and the Cisco 2600 and Cisco 3600 series routers: slot/port:ds0-group-no. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Examples
The following is sample output from the show voice port summary command for all voice ports on a Cisco MC3810 multiservice concentrator with an analog voice module (AVM):
PORT SIG-TYPE ADMIN OPER IN-STATUS OUT-STATUS GAIN ATTN CANCEL
1/1 fxs-ls up up on-hook idle 0 0 y
1/2 fxs-ls up up on-hook idle 0 0 y
1/3 e&m-wnk up up idle idle 0 0 y
1/4 e&m-wnk up up idle idle 0 0 y
1/5 fxo-ls up up idle on-hook 0 0 y
1/6 fxo-ls up up idle on-hook 0 0 y
The following is sample output from the show voice port summary command on a Cisco MC3810 multiservice concentrator with a digital voice module (DVM):
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:18 19 fxo-ls up dorm idle on-hook y
0:19 20 fxo-ls up dorm idle on-hook y
0:20 21 fxo-ls up dorm idle on-hook y
0:21 22 fxo-ls up dorm idle on-hook y
0:22 23 fxo-ls up dorm idle on-hook y
0:23 24 e&m-imd up dorm idle idle y
1/1 -- fxs-ls up dorm on-hook idle y
1/2 -- fxs-ls up dorm on-hook idle y
1/3 -- e&m-imd up dorm idle idle y
1/4 -- e&m-imd up dorm idle idle y
1/5 -- fxo-ls up dorm idle on-hook y
1/6 -- fxo-ls up dorm idle on-hook y
sys/voip/ccvpm vpm_htsp.c (107)
sys/voip/ccvtsp vtsp_core.c (167)
sys/voip/cli voiceport_action.c (58)
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US
The following is sample output from the show voice port command for an E&M analog voice port on a Cisco 3600:
E&M Slot is 1, Sub-unit is 0, Port is 0
Operation State is unknown
Administrative State is unknown
The Interface Down Failure Cause is 0
Noise Regeneration is disabled
Non Linear Processing is disabled
Music On Hold Threshold is Set to 0 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is disabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal
Initial Time Out is set to 0 s
Interdigit Time Out is set to 0 s
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Voice card specific Info Follows:
Signal Type is wink-start
Impedance is set to 600r Ohm
Digit Duration Timing is set to 0 ms
InterDigit Duration Timing is set to 0 ms
Pulse Rate Timing is set to 0 pulses/second
InterDigit Pulse Duration Timing is set to 0 ms
Clear Wait Duration Timing is set to 0 ms
Wink Wait Duration Timing is set to 0 ms
Wink Duration Timing is set to 0 ms
Delay Start Timing is set to 0 ms
Delay Duration Timing is set to 0 ms
The following is sample output from the show voice port command for a Foreign Exchange Station (FXS) analog voice port on a Cisco 3600:
Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0
Operation State is DORMANT
Administrative State is UP
The Interface Down Failure Cause is 0
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to 0 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 16ms
Connection Mode is Normal
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for northamerica
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Voice card specific Info Follows:
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Hook Flash Duration Timing is set to 600 ms
The following is sample output from the show voice port command for an FXS analog voice port on a Cisco MC3810 multiservice concentrator:
Voice port 1/2 Slot is 1, Port is 2
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Voice Activity Detection is disabled
Ringing Time Out is 180 s
Wait Release Time Out is 30 s
Nominal Playout Delay is 80 milliseconds
Maximum Playout Delay is 160 milliseconds
Region Tone is set for northamerica
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Analog interface A-D gain offset = -3 dB
Analog interface D-A gain offset = -3 dB
Voice card specific Info Follows:
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is active
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Ring Cadence are [20 40] * 100 msec
InterDigit Pulse Duration Timing is set to 500 ms
The following is sample output from the show voice port command for an E&M digital voice port on a Cisco 3600:
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US
The following is sample output from the show voice port summary command for all voice ports on a Cisco MC3810 multiservice concentrator with an analog voice module (AVM):
PORT SIG-TYPE ADMIN OPER IN-STATUS OUT-STATUS CODEC VAD GAIN ATTN CANCEL
1/1 fxs-ls up up on-hook idle 729a n 0 0 y
1/2 fxs-ls up up on-hook idle 729a n 0 0 y
1/3 e&m-wnk up up idle idle 729a n 0 0 y
1/4 e&m-wnk up up idle idle 729a n 0 0 y
1/5 fxo-ls up up idle on-hook 729a n 0 0 y
1/6 fxo-ls up up idle on-hook 729a n 0 0 y
Table 64 explains the fields in the sample output
Table 64 show voice port Field Descriptions
Field
|
Description
|
Administrative State
|
Administrative state of the voice port.
|
Alias
|
User-supplied alias for the voice port.
|
Analog interface A-D gain offset
|
Offset of the gain for analog-to-digital conversion.
|
Analog interface D-A gain offset
|
Offset of the gain for digital-to-analog conversion.
|
Clear Wait Duration Timing
|
Time of inactive seizure signal to declare call cleared.
|
Coder Type
|
Voice compression mode used.
|
Companding Type
|
Companding standard used to convert between analog and digital signals in PCM systems.
|
Connection Mode
|
Connection mode of the interface.
|
Connection Number
|
Full E.164 telephone number used to establish a connection with the trunk or PLAR mode.
|
Currently Processing
|
Type of call currently being processed: none, voice, or fax.
|
Delay Duration Timing
|
Maximum delay signal duration for delay dial signaling.
|
Delay Start Timing
|
Timing of generation of delayed start signal from detection of incoming seizure.
|
Description
|
Description of the voice port.
|
Dial Type
|
Out-dialing type of the voice port.
|
Digit Duration Timing
|
DTMF digit duration in milliseconds.
|
E&M Type
|
Type of E&M interface.
|
Echo Cancel Coverage
|
Echo cancel coverage for this port.
|
Echo Cancellation
|
Whether or not echo cancellation is enabled for this port.
|
Hook Flash Duration Timing
|
Maximum length of hook flash signal.
|
Hook Status
|
Hook status of the FXO/FXS interface.
|
Impedance
|
Configured terminating impedance for the E&M interface.
|
In Gain
|
Amount of gain inserted at the receiver side of the interface.
|
In Seizure
|
Incoming seizure state of the E&M interface.
|
Initial Time Out
|
Amount of time the system waits for an initial input digit from the caller.
|
InterDigit Duration Timing
|
DTMF interdigit duration, in milliseconds.
|
InterDigit Pulse Duration Timing
|
Pulse dialing interdigit timing, in milliseconds.
|
Interdigit Time Out
|
Amount of time the system waits for a subsequent input digit from the caller.
|
Maintenance Mode
|
Maintenance mode of the voice port.
|
Maximum Playout Delay
|
The amount of time before the Cisco MC3810 multiservice concentrator DSP starts to discard voice packets from the digital signal processor (DSP) buffer.
|
Music On Hold Threshold
|
Configured Music-On-Hold Threshold value for this interface.
|
Noise Regeneration
|
Whether or not background noise should be played to fill silent gaps if VAD is activated.
|
Nominal Playout Delay
|
The amount of time the Cisco MC3810 multiservice concentrator DSP waits before starting to play out the voice packets from the DSP buffer.
|
Non-Linear Processing
|
Whether or not nonlinear processing is enabled for this port.
|
Number of signaling protocol errors
|
Number of signaling protocol errors.
|
Operations State
|
Operation state of the port.
|
Operation Type
|
Operation of the E&M signal: two-wire or four-wire.
|
Out Attenuation
|
Amount of attenuation inserted at the transmit side of the interface.
|
Out Seizure
|
Outgoing seizure state of the E&M interface.
|
Port
|
Port number for this interface associated with the voice interface card.
|
Pulse Rate Timing
|
Pulse dialing rate, in pulses per second (pps).
|
Region Tone
|
Configured regional tone for this interface.
|
Ring Active Status
|
Ring active indication.
|
Ring Cadence
|
Configured ring cadence for this interface.
|
Ring Frequency
|
Configured ring frequency for this interface.
|
Ring Ground Status
|
Ring ground indication.
|
Ringing Time Out
|
Ringing timeout duration.
|
Signal Type
|
Type of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial.
|
Slot
|
Slot used in the voice interface card for this port.
|
Sub-unit
|
Subunit used in the voice interface card for this port.
|
Tip Ground Status
|
Tip ground indication.
|
Type of VoicePort
|
Type of voice port: FXO, FXS, or E&M.
|
The Interface Down Failure Cause
|
Text string describing why the interface is down,
|
Voice Activity Detection
|
Whether Voice Activity Detection is enabled or disabled.
|
Wait Release Time Out
|
The length of time a voice port stays in the call-failure state while the Cisco MC3810 multiservice concentrator sends a busy tone, a reorder tone, or an out-of-service tone to the port.
|
Wink Duration Timing
|
Maximum wink duration for wink start signaling.
|
Wink Wait Duration Timing
|
Maximum wink wait duration for wink start signaling.
|
The following example displays voice port configuration information for the digital voice port 0 located in slot 1, DS0 group 1:
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 DBMS
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US
The following is sample output from the Cisco AS5800 for the show voice port command:
Type of VoicePort is ISDN
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 16 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Region Tone is set for US
Table 65 describes the significant fields shown in the display.
Table 65 show voice port Field Descriptions for the Cisco AS5800
Field
|
Description
|
Type of VoicePort
|
Indicates the voice port type.
|
Operational State
|
Operational state of the voice port.
|
Administrative State
|
Administrative state of the voice port.
|
Clear Wait Duration Timing
|
Time of inactive seizure signal to declare call cleared.
|
Currently Processing
|
Type of call currently being processed: none, voice, or fax.
|
Operations State
|
Operation state of the port.
|
Operation Type
|
Operation of the E&M signal: two-wire or four-wire.
|
Noise Regeneration
|
Whether or not background noise should be played to fill silent gaps if VAD is activated.
|
Non-Linear Processing
|
Whether or not nonlinear processing is enabled for this port.
|
Music-On-Hold Threshold
|
Configured music-on-hold threshold value for this interface.
|
In Gain
|
Amount of gain inserted at the receiver side of the interface.
|
Out Attenuation
|
Amount of attenuation inserted at the transmit side of the interface.
|
Pulse Rate Timing
|
Pulse dialing rate, in pulses per second (pps).
|
Echo Cancellation
|
Whether or not echo cancellation is enabled for this port.
|
Echo Cancel Coverage
|
Echo cancel coverage for this port.
|
Connection Mode
|
Connection mode of the interface.
|
Connection Number
|
Full E.164 telephone number used to establish a connection with the trunk or PLAR mode.
|
Initial Time Out
|
Amount of time the system waits for an initial input digit from the caller.
|
Interdigit Time Out
|
Amount of time the system waits for a subsequent input digit from the caller.
|
Regional Tone
|
Configured regional tone for this interface.
|
show voice trunk-conditioning signaling
To display the status of trunk-conditioning signaling and timing parameters for a voice port, use the show voice trunk-conditioning signaling command in user EXEC or privileged EXEC mode.
show voice trunk-conditioning signaling [summary | voice-port]
Syntax Description
summary
|
(Optional) Displays a summary of the status for all voice ports on the router or concentrator.
|
voice-port
|
(Optional) Displays a detailed report for a specified voice port.
|
Command Modes
User EXEC
Privileged EXEC
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810 multiservice concentrator as show voice permanent-call.
|
12.0(4)T
|
This command was integrated into the 12.0(4)T release.
|
12.0(7)XK
|
This command was renamed show voice trunk-conditioning signaling.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
The show voice trunk-conditioning signaling command displays the trunk signaling status for analog and digital voice ports on Cisco MC3810 multiservice concentrators and Cisco 2600 and 3600 series routers.
Examples
The following is a sample display from the show voice trunk-conditioning signaling summary command for voice ports on a Cisco MC3810 multiservice concentrator:
Router# show voice trunk-conditioning signaling summary
TX INFO :slow-mode seq#= 25, sig pkt cnt= 40, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS signal path is OPEN
RX INFO :slow-mode, sig pkt cnt= 36, prev-seq#= 25, last-ABCD=0000
The following is a sample display from the show voice trunk-conditioning signaling command for voice port 1/5 on a Cisco MC3810 multiservice concentrator:
Router# show voice trunk-conditioning signaling 1/5
TX INFO :slow-mode seq#= 25, sig pkt cnt= 42, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
RX INFO :slow-mode, sig pkt cnt= 37
missing = 0, out of seq = 0, very late = 0
playout depth = 0 (ms), refill count = 1
prev-seq#= 25, last-ABCD=0000
trunk_down_timer = 4212 (ms), idle timer = 0 (sec),
tx_oos_timer = 0 (sec), rx_ais_duration = 0 (ms)
forced playout signal pattern = NONE
signaling playout history
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
The following is a sample display from the show voice trunk-conditioning signaling summary command for voice ports on a Cisco 3600 series router:
Router# show voice trunk-conditioning signaling summary
The following is a sample display from the show voice trunk-conditioning signaling command for voice port 3/0:6 on a Cisco 3600 series router:
Router# show voice trunk-conditioning signaling 3/0:6
hardware-state ACTIVE signal type is NorthamericanCAS
forced playout pattern = STOPPED
trunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0
Table 66 describes the significant fields shown in the display.
Table 66 show voice trunk-conditioning signaling Field Descriptions
Field
|
Description
|
current timer
|
Time since last signaling packets were received.
|
forced playout pattern
|
Which forced playout pattern is sent to PBX:
• 0 = no forced playout pattern is sent
• 1 = receive IDLE playout pattern is sent
• 2 = receive OOS playout pattern is sent
|
hardware-state
|
Hardware state based on received IDLE pattern:
IDLE = both sides are idle
ACTIVE = at least one side is active
|
signal type
|
Signaling type used by lower level driver: northamerica, melcas, transparent, or external.
|
idle timer
|
Time the hardware on both sides has been in idle state.
|
last-ABCD
|
Last received or transmitted signal bit pattern.
|
max inter-arrival time
|
Maximum interval between received signaling packets.
|
missing
|
Number of missed signal packets.
|
mode
|
Signaling packet generation frequency:
• fast mode = every 4 milliseconds
• slow mode = same frequency as keepalive timer
|
out of seq
|
Number of out-of-sequence signal packets.
|
playout depth
|
Number of packets in playout buffer.
|
prev-seq#
|
Sequence number of previous signaling packet.
|
refill count
|
Number of packets created to maintain nominal length of playout packet buffer.
|
rx_ais_duration
|
Time since receipt of AIS indicator.
|
seq#
|
Sequence number of signaling packet.
|
sig pkt cnt
|
Number of transmitted or received signaling packets.
|
signal path
|
Status of signaling path.
|
signaling playout history
|
Signaling bits received in last 60 milliseconds.
|
trunk_down_timer
|
Time since last signaling packets were received.
|
tx_oos_timer
|
Time since PBX started sending OOS signaling pattern defined by signal pattern oos transmit.
|
very late
|
Number of very late signaling packets.
|
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays the configuration for all VoIP and POTS dial peers configured on the router.
|
show voice dsp
|
Shows the current status of all DSP voice channels.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show voice trunk-conditioning supervisory
|
Displays the status of trunk supervision and configuration parameters for voice ports.
|
show voice trunk-conditioning supervisory
To display the status of trunk supervision and configuration parameters for a voice port, use the show voice trunk-conditioning supervisory command in user EXEC or privileged EXEC mode.
show voice trunk-conditioning supervisory [summary | voice-port]
Syntax Description
summary
|
(Optional) Displays a summary of the status for all voice ports on the router or concentrator.
|
voice-port
|
(Optional) Displays a detailed report for a specified voice port.
|
Command Modes
User EXEC
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 2600 and 3600 series routers, and the MC3810 multiservice concentrator.
|
12.1(2)T
|
This command was integrated into the 12.1(2)T release.
|
12.1(3)T
|
This command was first supported on the Cisco 2600 and 3600 series routers.
|
Usage Guidelines
The show voice trunk-conditioning supervisory command displays the trunk supervision and configuration status for analog and digital voice ports.
Examples
The following is a sample display from the show voice trunk-conditioning supervisory summary command for voice ports on a Cisco MC3810 multiservice concentrator:
Router# show voice trunk-conditioning supervisory summary
1/5 : state : TRUNK_SC_CONNECT, voice : on , signal : on ,slave
The following is a sample display from the show voice trunk-conditioning supervisory command for voice port 1/5 on a Cisco MC3810 multiservice concentrator:
Router# show voice trunk-conditioning supervisory 1/5
1/5 : state : TRUNK_SC_CONNECT, voice : on, signal : on, slave
sequence oos : idle and oos
pattern :rx_idle = 0x0 rx_oos = 0xF tx_oos = 0xF
timing : idle = 0, restart = 0, standby = 0, timeout = 40
supp_all = 50, supp_voice = 0, keep_alive = 5
timer: oos_ais_timer = 0, timer = 0
The following is a sample display from the show voice trunk-conditioning supervisory summary command for voice ports on a Cisco 3600 series router:
Router# show voice trunk-conditioning supervisory summary
3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/0:7(7) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:0(8) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:1(1) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:3(3) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
The following is a sample display from the show voice trunk-conditioning supervisory command for voice port 3/0:6 on a Cisco 3600 series router:
Router# show voice trunk-conditioning supervisory 3/0:6
3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on, signal : on, master
sequence oos : idle and oos
pattern :rx_idle = 0x0 rx_oos = 0xF
timing : idle = 0, restart = 0, standby = 0, timeout = 40
supp_all = 0, supp_voice = 0, keep_alive = 5
timer: oos_ais_timer = 0, timer = 0
Table 67 describes the significant fields shown in the display.
Table 67 show voice trunk-conditioning supervisory Field Descriptions
Field
|
Description
|
keep_alive
|
Signaling packets periodically sent to the far end, even if there is no signal change. These signaling packets function as keep alive messages.
|
master
|
The voice port configured as "connect trunk xxxx."
|
slave
|
The voice port configured as "connect trunk xxxx answer-mode."
|
oos_ais_timer
|
Time since the signaling packet with AIS indicator was received.
|
pattern
|
4-bit signaling pattern.
|
restart
|
The restart timeout after far end is OOS.
|
rx-idle
|
The signaling bit pattern indicating that the far end is idle.
|
rx-oos
|
The signaling bit pattern sent to the PBX indicating that the network is OOS.
|
standby
|
The time before the slave side goes back to standby after the far end goes OOS.
|
supp_all
|
The timeout before suppressing transmission of voice and signaling packets to the far end after detection of PBX OOS.
|
supp_voice
|
The timeout before suppressing transmission of voice packet to the far end after detection of PBX oos.
|
timeout
|
The timeout for nonreceipt of keepalive packets before the far end is considered to be OOS.
|
TRUNK_SC_CONNECT
|
Trunk conditioning supervisory component status.
|
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays the configuration for all VoIP and POTS dial peers configured on the router.
|
show voice dsp
|
Displays the current status of all DSP voice channels.
|
show voice port
|
Displays configuration information about a specific voice port.
|
show voice trunk-conditioning signaling
|
Displays the status of trunk-conditioning signaling and timing parameters for a voice port
|
show vrm active_calls
To display active-only voice calls either for a specific voice feature card (VFC) or for all VFCs, use the show vrm active_calls command in privileged EXEC mode.
show vrm active_calls {dial-shelf-slot-number | all}
Syntax Description
dial-shelf-slot-number
|
Slot number of the dial shelf. Valid number is 0 to 13.
|
all
|
Lists all active calls for VFC slots.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5800 universal access server.
|
Usage Guidelines
Use the show vrm active_calls to display active-only voice calls either for a specific VFC or for all VFCs. Each active call occupies a block of information describing the call. This information provides basically the same information as the show vrm vdevice command.
Examples
The following is sample output from the show vrm active_calls command specifying dial shelf slot number:
Router# show vrm active_calls 6
slot = 6 virtual voice dev (tag) = 61 channel id = 2
capabilities list map = 9FFF
last/current codec loaded/used = None
Resource (vdev_common) status = 401 means :active others
tot ingress control = 1308
tot ingress data drops = 0
tot ingress control drops = 0
tot egress control = 1304
tot egress data drops = 0
tot egress control drops = 0
slot = 6 virtual voice dev (tag) = 40 channel id = 2
capabilities list map = 9FFF
last/current codec loaded/used = None
Resource (vdev_common) status = 401 means :active others
Table 68 describes the significant fields shown in the display.
Table 68 show vrm active_calls Field Descriptions
Field
|
Description
|
slot
|
Slot where voice card is installed.
|
virtual voice dev (tag)
|
Identification number of the virtual voice device.
|
channel id
|
Identification number of the channel associated with this virtual voice device.
|
capability list map
|
Bitmaps for the codec supported on that DSP channel. Available values are:
CC_CAP_CODEC_G711U: 0x1
CC_CAP_CODEC_G711A: 0x2
CC_CAP_CODEC_G729IETF: 0x4
CC_CAP_CODEC_G729a: 0x8
CC_CAP_CODEC_G726r16: 0x10
CC_CAP_CODEC_G726r24: 0x20
CC_CAP_CODEC_G726r32: 0x40
CC_CAP_CODEC_G728: 0x80
CC_CAP_CODEC_G723r63: 0x100
CC_CAP_CODEC_G723r53: 0x200
CC_CAP_CODEC_GSM: 0x400
CC_CAP_CODEC_G729b: 0x800
CC_CAP_CODEC_G729ab: 0x1000
CC_CAP_CODEC_G723ar63: 0x2000
CC_CAP_CODEC_G723ar53: 0x4000
CC_CAP_CODEC_G729: 0x8000
|
last/current codec loaded/used
|
Indicates the last codec loaded or used.
|
TDM time slot
|
Time division multiplexing time slot.
|
Resource (vdev_common) status
|
Current status of the VFC.
|
tot ingress data
|
Total amount of data (number of packets) sent from the PSTN side of the connection to the VoIP side of the connection.
|
tot ingress control
|
Total number of control packets sent from the PSTN side of the connection to the Voice over IP (VoIP) side of the connection.
|
tot ingress data drops
|
Total number of data packets dropped from the PSTN side of the connection to the VoIP side of the connection.
|
tot ingress control drops
|
Total number of control packets dropped from the PSTN side of the connection to the VoIP side of the connection.
|
tot egress data
|
Total amount of data (number of packets) sent from the VoIP side of the connection to the PSTN side of the connection.
|
tot egress control
|
Total number of control packets sent from the VoIP side of the connection to the PSTN side of the connection.
|
tot egress data drops
|
Total number of data packets dropped from the VoIP side of the connection to the PSTN side of the connection.
|
tot egress control drops
|
Total number of control packets dropped from the VoIP side of the connection to the PSTN side of the connection.
|
Related Commands
Command
|
Description
|
show vrm vdevices
|
Displays detailed information for a specific DSP or a brief summary display for all VFCs.
|
show vrm vdevices
To display detailed information for a specific digital signal processor (DSP) or a brief summary display for all voice feature cards (VFCs), use the show vrm vdevices command in privileged EXEC mode.
show vrm vdevices {{vfc-slot-number | voice-device-number} | summary}
Syntax Description
vfc-slot-number
|
Slot number of the VFC. Valid number is from 0 to 11.
|
voice-device-number
|
DSP number. Valid number is from 1 to 96.
|
summary
|
List synopsis of voice feature card DSP mappings, capabilities, and resource states.
|
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.0(7)T
|
This command was introduced on the Cisco AS5800 universal access server.
|
Usage Guidelines
Use the show vrm vdevices command to display detailed information for a specific DSP or a brief summary display for all VFCs. The display provides information on the number of channels, channels per DSP, bitmap of DSPMs, version numbers, and so on. This information is useful in monitoring the current state of your VFCs.
The display for a specific DSP provides information on the codec that each channel is using, if active, or last used and if the channel is not currently sending cells. It also displays the state of the resource. In most cases, if there is an active call on that channel, the resource should be marked active. If the resource is marked as reset or bad, this may be an indication of a response loss for the VFC on a reset request. If this condition persists, you might experience a problem with the communication link between the router shelf and the VFC.
Examples
The following is sample output from the show vrm vdevices command specifying dial shelf slot number and DSP number. In this particular example, the call is active so the statistics displayed are for this active call. If no calls are currently active on the device, the statistics would be for the previous (or last active) call.
Router# show vrm vdevices 6 1
slot = 6 virtual voice dev (tag) = 1 channel id = 1
capabilities list map = 9FFF
last/current codec loaded/used = None
Resource (vdev_common) status = 401 means :active others
tot ingress control = 1194
tot ingress data drops = 0
tot ingress control drops = 0
tot egress control = 1209
tot egress data drops = 0
tot egress control drops = 0
slot = 6 virtual voice dev (tag) = 1 channel id = 2
capabilities list map = 9FFF
last/current codec loaded/used = None
Resource (vdev_common) status = 401 means :active others
tot ingress control = 1167
tot ingress data drops = 0
tot ingress control drops = 0
tot egress control = 1163
tot egress data drops = 0
tot egress control drops = 0
Table 69 describes the significant fields shown in the display.
Table 69 show vrm vdevices Field Descriptions
Field
|
Description
|
slot
|
Slot where voice card is installed.
|
virtual voice dev (tag)
|
Identification number of the virtual voice device.
|
channel id
|
Identification number of the channel associated with this virtual voice device.
|
capability list map
|
Bitmaps for the codec supported on that DSP channel. Available values are:
• CC_CAP_CODEC_G711U: 0x1
• CC_CAP_CODEC_G711A: 0x2
• CC_CAP_CODEC_G729IETF: 0x4
• CC_CAP_CODEC_G729a: 0x8
• CC_CAP_CODEC_G726r16: 0x10
• CC_CAP_CODEC_G726r24: 0x20
• CC_CAP_CODEC_G726r32: 0x40
• CC_CAP_CODEC_G728: 0x80
• CC_CAP_CODEC_G723r63: 0x100
• CC_CAP_CODEC_G723r53: 0x200
• CC_CAP_CODEC_GSM: 0x400
• CC_CAP_CODEC_G729b: 0x800
• CC_CAP_CODEC_G729ab: 0x1000
• CC_CAP_CODEC_G723ar63: 0x2000
• CC_CAP_CODEC_G723ar53: 0x4000
• CC_CAP_CODEC_G729: 0x8000
|
last/current codec loaded/used
|
The last codec loaded or used.
|
TDM time slot
|
Time division multiplexing time slot.
|
Resource (vdev_common) status
|
Current status of the VFC. Possible field values are:
• FREE = 0x0000
• ACTIVE_CALL = 0x0001
• BUSYOUT_REQ = 0x0002
• BAD = 0x0004
• BACK2BACK_TEST = 0x0008
• RESET = 0x0010
• DOWNLOAD_FILE = 0x0020
• DOWNLOAD_FAIL = 0x0040
• SHUTDOWN = 0x0080
• BUSY = 0x0100
• OIR = 0x0200
• HASLOCK = 0x0400 /* vdev_pool has locked port */
• DOWNLOAD_REQ = 0x0800
• RECOVERY_REQ = 0x1000
• NEGOTIATED = 0x2000
• OOS = 0x4000
|
tot ingress data
|
Total amount of data (number of packets) sent from the PSTN side of the connection to the Voice over IP (VoIP) side of the connection.
|
tot ingress control
|
Total number of control packets sent from the PSTN side of the connection to the VoIP side of the connection.
|
tot ingress data drops
|
Total number of data packets dropped from the PSTN side of the connection to the VoIP side of the connection.
|
tot ingress control drops
|
Total number of control packets dropped from the PSTN side of the connection to the VoIP side of the connection.
|
tot egress data
|
Total amount of data (number of packets) sent from the VoIP side of the connection to the PSTN side of the connection.
|
tot egress control
|
Total number of control packets sent from the VoIP side of the connection to the PSTN side of the connection.
|
tot egress data drops
|
Total number of data packets dropped from the VoIP side of the connection to the PSTN side of the connection.
|
tot egress control drops
|
Total number of control packets dropped from the VoIP side of the connection to the PSTN side of the connection.
|
The following is sample output from the show vrm devices command specifying a summary list. In the "Voice Device Mapping" area, the "C_Ac" column indicates number of active calls for a specific DSP. If there are any nonzero numbers under the "C_Rst" and/or "C_Bad" column, this indicates that a reset request was sent but it was lost; this could mean a faulty DSP.
Router# show vrm vdevices summary
***********************************************************
******summary of voice devices for all voice cards*********
***********************************************************
slot = 6 major ver = 0 minor ver = 1 core type used = 2
number of modules = 16 number of voice devices (DSPs) = 96
chans per vdevice = 2 tot chans = 192 tot active calls = 178
module presense bit map = FFFF tdm mode = 1 num_of_tdm_timeslots = 384
number of default voice file (core type images) = 2
file 0 maj ver = 0 min ver = 0 core_type = 1
trough size = 2880 slop value = 0 built-in codec bitmap = 0
loadable codec bitmap = 0 fax codec bitmap = 0
file 1 maj ver = 3 min ver = 1 core_type = 2
trough size = 2880 slop value = 1440 built-in codec bitmap = 40B
loadable codec bitmap = BFC fax codec bitmap = 7E
-------------------Voice Device Mapping------------------------
Logical Device (Tag) Module# DSP# C_Ac C_Busy C_Rst C_Bad
---------------------------------------------------------------
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Total active call channels = 178
Total busied out channels = 0
Total channels in reset = 0
Note :Channels could be in multiple states
Table 70 describes the significant fields shown in the display.
Table 70 show vrm vdevices summary Field Descriptions
Field
|
Description
|
slot
|
Slot number where VFC is installed.
|
major ver
|
Major version of firmware running on VFC.
|
minor ver
|
Minor version of firmware running on VFC.
|
core type used
|
Type of DSPware in use. Possible field values are:
• 1 = UBL (boot loader)
• 2 = high complexity core
• 3 = medium complexity core
• 4 = low complexity core
• 255 = invalid
|
number of modules
|
Number of modules on the VFC. Maximum is 16.
|
number of voice devices (DSP)s
|
Number of possible DSPs. Maximum number is 96.
|
chans per vdevice
|
Number of channels (meaning calls) each DSP can handle.
|
tot chans
|
Total number of channels.
|
tot active calls
|
Total number of active calls on this VFC.
|
module presense bit map
|
Indicates a 16-bit bitmap, each bit representing a module.
|
tdm mode
|
Time division multiplex bus mode. Possible field values are:
• 0 = VFC is in classic mode.
• 1 = VFC is in plus mode.
This field should always be 1.
|
num_of_tdm_time slots
|
Total number of calls that can be handled by the VFC.
|
auto recovery
|
Indicates whether auto recovery is enabled. When autorecovery is enabled, the VRM will try to recover a DSP by resetting it if, for some reason, the DSP stops responding.
|
number of default voice file (core type images)
|
Number of DSPware files in use.
|
maj ver
|
Major version of the DSPware in use.
|
min ver
|
Minor version of the DSPware in use.
|
core type
|
Type of DSPware in use: Possible field values are:
• 1 = boot loader
• 2 = high complexity core
• 3 = medium complexity core
• 4 = low complexity core
|
trough size
|
This value indirectly represents the complexity of the DSPware in use.
|
slop value
|
This value indirectly represents the complexity of the DSPware in use.
|
built-in codec bitmap
|
Represents the bitmap of the codec built into the DSP firmware. Possible field values are:
• CC_CAP_CODEC_G711U 0x0001
• CC_CAP_CODEC_G711A 0x0002
• CC_CAP_CODEC_G729IETF 0x0004
• CC_CAP_CODEC_G729a 0x0008
• CC_CAP_CODEC_G726r16 0x0010
• CC_CAP_CODEC_G726r24 0x0020
• CC_CAP_CODEC_G726r32 0x0040
• CC_CAP_CODEC_G728 0x0080
• CC_CAP_CODEC_G723r63 0x0100
• CC_CAP_CODEC_G723r53 0x0200
• CC_CAP_CODEC_GSM 0x0400
• CC_CAP_CODEC_G729b 0x0800
• CC_CAP_CODEC_G729ab 0x1000
• CC_CAP_CODEC_G723ar63 0x2000
• CC_CAP_CODEC_G723ar53 0x4000
• CC_CAP_CODEC_G729 0x8000
|
loadable codec bitmap
|
Represents the loadable codec bitmap for the loadable codecs. Possible field values are:
• CC_CAP_CODEC_G711U = 0x0001
• CC_CAP_CODEC_G711A = 0x0002
• CC_CAP_CODEC_G729IETF = 0x0004
• CC_CAP_CODEC_G729a = 0x0008
• CC_CAP_CODEC_G726r16 = 0x0010
• CC_CAP_CODEC_G726r24 = 0x0020
• CC_CAP_CODEC_G726r32 = 0x0040
• CC_CAP_CODEC_G728 = 0x0080
• CC_CAP_CODEC_G723r63 = 0x0100
• CC_CAP_CODEC_G723r53 = 0x0200
• CC_CAP_CODEC_GSM = 0x0400
• CC_CAP_CODEC_G729b = 0x0800
• CC_CAP_CODEC_G729ab = 0x1000
• CC_CAP_CODEC_G723ar63 = 0x2000
• CC_CAP_CODEC_G723ar53 = 0x4000
• CC_CAP_CODEC_G729 = 0x8000
|
fax codec bitmap
|
Represents the fax codec bitmap. Possible field values are:
• FAX_NONE = 0x1
• FAX_VOICE = 0x2
• FAX_144 = 0x4
• FAX_96 = 0x8
• FAX_72 = 0x10
• FAX_48 = 0x20
• FAX_24 = 0x40
|
Logical Device (Tag)
|
Tag number or DSP number on that VFC.
|
Module #
|
Number identifying the module associated with a specific logical device.
|
DSP#
|
Number identifying the DSP on the VFC.
|
C_Ac
|
Number of active calls on identified DSP.
|
C_Busy
|
Number of busied-out channels associated with identified DSP.
|
C_Rst
|
Number of channels in the reset state associated with identified DSP.
|
C_Bad
|
Number of defective ("bad") channels associated with identified DSP.
|
Total active call channels
|
Total number of active calls.
|
Total busied out channels
|
Total number of busied-out channels.
|
Total channels in reset
|
Total number of channels in reset state.
|
Total bad channels
|
Total number of defective channels.
|
Related Commands
Command
|
Description
|
show vrm active_calls
|
Displays active-only voice calls for either a specific VFC or all VFCs.
|
shut
To shut down a set of digital signal processors (DSPs) on the Cisco 7200 series router, use the shut command in DSP configuration mode. To put DSPs back in service, use the no form of this command.
shut number
no shut number
Syntax Description
number
|
Number of DSPs to be shut down.
|
Defaults
no shut
Command Modes
DSP configuration
Command History
Release
|
Modification
|
12.0(5)XE
|
This command was introduced on the Cisco 7200 series router.
|
12.1(1)T
|
This command was modified to add information about DSP groups.
|
Usage Guidelines
This command applies to Voice over IP on the Cisco 7200 series routers.
Examples
The following example shuts down two sets of DSPs:
shutdown (dial-peer)
To change the administrative state of the selected dial peer from up to down, use the shutdown command in dial-peer configuration mode. To change the administrative state of this dial peer from down to up, use the no form of this command.
shutdown
no shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
no shutdown
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
12.1(1)
|
This command was modified for store-and-forward fax.
|
Usage Guidelines
When a dial peer is shut down, you cannot initiate calls to that peer.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example changes the administrative state of voice telephony (plain old telephone service [POTS]) dial peer 10 to down:
The following example changes the administrative state of voice telephony (POTS) dial peer 10 to up:
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode, defines the type of dial peer, and defines the dial-peer tag number.
|
shutdown (DS1)
To shut down a DS1 link (send a Blue Alarm), use the shutdown command in controller configuration mode. To activate the DS1 (cancel the sending of the Blue Alarm), use the no form of the command.
shutdown
no shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
no shutdown
Command Modes
Controller configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced.
|
Usage Guidelines
This command applies to Voice over Frame Relay and Voice over ATM on the Cisco MC3810 multiservice concentrator.
Examples
The following example shuts down a DS1 link on controller T1 0:
shutdown (gatekeeper)
To disable the gatekeeper, use the shutdown command in gatekeeper configuration mode. To enable the gatekeeper, use the no form of this command.
shutdown
no shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled (shut down)
Command Modes
gatekeeper configuration
Command History
Release
|
Modification
|
11.3(2)NA
|
This command was introduced on the Cisco 2500 series and Cisco 3600 series routers.
|
12.0(3)T
|
The command was integrated into Cisco IOS Release 12.0(3)T and supported on the Cisco MC3810 multiservice concentrator.
|
Usage Guidelines
The gatekeeper does not have to be enabled before you can use the other gatekeeper configuration commands. In fact, it is recommended that you complete the gatekeeper configuration before bringing up the gatekeeper because some characteristics may be difficult to alter while the gatekeeper is running, as there may be active registrations or calls.
The no shutdown command enables the gatekeeper, but it does not make the gatekeeper operational. The two exceptions to this are as follows:
•
If no local zones are configured, a no shutdown command places the gatekeeper in INACTIVE mode waiting for a local zone definition.
•
If local zones are defined to use an HSRP virtual address, and the HSRP interface is in STANDBY mode, the gatekeeper goes into HSRP STANDBY mode. Only when the HSRP interface is ACTIVE will the gatekeeper go into the operational UP mode.
Examples
The following command disables a gatekeeper:
shutdown (RLM)
To shut down all of the links under the RLM group, use the shutdown command in RLM configuration mode. RLM will not try to reestablish those links until the command is negated. To disable this function, use the no form of this command.
shutdown
no shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
RLM configuration
Command History
Release
|
Modification
|
11.3(7)
|
This command was introduced.
|
Related Commands
Command
|
Description
|
clear rlm group
|
Clears all RLM group time stamps to zero.
|
clear interface
|
Resets the hardware logic on an interface.
|
interface
|
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
|
link (RLM)
|
Specifies the link preference.
|
protocol rlm port
|
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
|
retry keepalive
|
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
|
server (RLM)
|
Defines the IP addresses of the server.
|
show rlm group statistics
|
Displays the network latency of the RLM group.
|
show rlm group status
|
Displays the status of the RLM group.
|
show rlm group timer
|
Displays the current RLM group timer values.
|
timer
|
Overwrites the default setting of timeout values.
|
shutdown (settlement)
To deactivate the settlement provider, use the shutdown command in settlement configuration mode. To activate a settlement provider, use the no shutdown command
shutdown
no shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
The default status of a settlement provider is deactivated. The settlement provider is down.
Command Modes
Settlement configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced the Cisco 2500 series and Cisco 3600 series routers and on the Cisco AS5300 universal access server.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
Use the no shutdown command at the end of the configuration of a settlement server to bring up the provider. This command activates the provider. Otherwise, transactions will not go through the provider to be audited and charged. Use the shutdown command to deactivate the provider.
Examples
The following example enables a settlement server:
The following example disables a settlement server:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the time that a connection is maintained after completing a communication exchange.
|
customer-id
|
Identifies a carrier or ISP with a settlement provider.
|
device-id
|
Specifies a gateway associated with a settlement provider.
|
encryption
|
Sets the encryption method to be negotiated with the provider.
|
max-connection
|
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
|
response-timeout
|
Configures the maximum time to wait for a response from a server.
|
retry-delay
|
Sets the time between attempts to connect with the settlement provider.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
settlement
|
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
shutdown (voice-port)
To take the voice ports for a specific voice interface card offline, use the shutdown command in voice-port configuration mode. To put the ports back in service, use the no form of this command.
shutdown
no shutdown
Syntax Description
This command has no arguments or keywords.
Defaults
shutdown
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 3600 series router.
|
Usage Guidelines
When you enter the shutdown command, all ports on the voice interface card are disabled. When you enter the no shutdown command, all ports on the voice interface card are enabled. A telephone connected to an interface will hear dead silence when a port is shut down.
Examples
The following example takes voice port 1/1/0 on the Cisco 3600 series offline:
Note
The preceding configuration example shuts down both voice ports 1/1/0 and 1/1/1.