Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2
Cisco IOS Voice, Video, and Fax Commands: R through Sh
Downloads: This chapterpdf (PDF - 2.14MB) The complete bookPDF (PDF - 8.06MB) | Feedback

Cisco IOS Voice, Video, and Fax Commands: R Through Sh

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:

dial-peer voice 111 pots
 port 1/0/0
 destination-pattern 5551212
 register e164

The following commands deregister an address with the gatekeeper:

dial-peer voice 111 pots
no register e164

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:

dial-peer voice 222 pots
port 1/0/0
destination 919555....
register e164


ERROR-register-e164:Dial-peer destination-pattern is not a full E.164 number

no gateway
dial-peer voice 111 pots
register e164


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.

pots country jp
dial-peer voice 1 pots
 registered-caller ring 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:

dial-peer voice 10 voip
 req-qos guaranteed-delay

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:

reset 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.

settlement 0
 response-timeout 1

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:

settlement 0
 relay-delay 15

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:

settlement 0
 retry-limit 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.
sip-ua
 retry invite 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:

ring 1

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:

voice-port 1/1
 ring cadence pattern02

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:

voice-port 1/2
 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:

voice-port 1/0/0
 ring cadence pattern04

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:

voice-port 1/0/0
 ring frequency 25

The following example configures the ring frequency on the Cisco MC3810 multiservice concentrator for 20 Hz:

voice-port 1/1
 ring frequency 20

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:

voice-port 1/0/0
 ring number 5

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:

voice-port 1/1
 ring number 5

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:

dial-peer voice 10 voip
roaming

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:

settlement 0
roaming 

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.

translation-rule 21
 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.

translation-rule 345
 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:

security h323id

The following example enables security, authenticating all users by using their H.323-IDs and a password of qwerty2x:

security h323-id
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 h323-id
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 e164
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:

dial-peer voice 200 vofr
 no sequence-numbers

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):

Gatekeeper
  server trigger arq sj.xyz.com 1 Server-123 1.14.93.130 1751
    destination-info e164 1800.......
    exit

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:

dial-peer voice 102 voip
 session protocol sipv2

The following example selects AAL2 trunking as the session protocol on a Cisco MC3810 multiservice concentrator:

dial-peer voice 10 voatm
 session protocol aal2-trunk

The following example selects Cisco Session Protocol as the session protocol on a Cisco 3600 series router:

dial-peer voice 20 voip
 session protocol cisco

The following example selects SMTP as the session protocol:

dial-peer voice 10 mmoip
 session protocol smtp

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:

dial-peer voice 200 vofr
 session protocol frf11-trunk
 called-number 5552150

The following example shows how to configure the FRF.11 session protocol on a Cisco MC3810 multiservice concentrator for VoFR dial peer 200:

dial-peer voice 200 vofr
 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:

voice service voatm
  session protocol aal2

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:

dial-peer voice 111 voip
 destination-pattern 111
 session protocol multicast
 session target ipv4:237.111.0.111:22222
 ip precedence 5
 codec g711ulaw

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.

dial-peer voice 12 voatm
 destination-pattern 13102221111
 session target atm0 pvc 20

The following example delivers fax-mail to multiple recipients:

dial-peer voice 10 mmoip
 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:

marketing-information: 
 john@example.com, 

 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.

dial-peer voice 12 voatm
 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:

dial-peer voice 11 vofr
 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:

dial-peer voice 200 vofr
 destination-pattern 13102221111
 called-number 5552150
 session protocol frf11-trunk
 session target serial 1/0 100 20

The following example delivers fax-mail to multiple recipients:

dial-peer voice 10 mmoip
 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:

marketing-information: 
 john@example.com, 

 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:

dial-peer voice 10 voip
 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."

dial-peer voice 10 voip
 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.

dial-peer voice 10 voip
 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.

dial-peer voice 10 voip
 destination-pattern 12345
 session target dns:$e$.cisco.com

The following example configures a session target using RAS:

dial-peer voice 11 voip
 destination-pattern 13102221111
 session target ras

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:

dial-peer voice 102 voip
 session transport udp

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:

dial-peer voice 10 voip
 destination-pattern 1408.......
 session target ipv4:172.22.95.14
 settle-call 0 

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:

settlement 0

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:

sgcp

The following example shows the SGCP daemon being disabled:

no sgcp

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
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:

sgcp graceful-shutdown

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:

sgcp modem passthru ca

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:

sgcp modem passthru nse
sgcp tse payload 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 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:

sgcp request retries 10

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:

sgcp request timeout 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 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:

sgcp restart delay 40

The following example shows the system configured to send an RSIP notification to the SGCP call agent when the T1 controller state changes:

sgcp restart notify

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:

sgcp timer rtp-nse 1000

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:

sgcp modem passthru nse
sgcp tse payload 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
   Successful Fail-Overs      :0  
   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
Session-Group
   Group Name :grp1
   Set Name   :set1
   Sessions   :3
    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
    RUDP Options
      timer cumulative ack :100
      timer keepalive      :1000
      timer retransmit     :300
      timer transfer state :2000
      receive max          :32
      cumulative ack max   :3
      retrans max          :2
      out-of-sequence max  :3
      auto-reset max       :5

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
Session-Group
Group Name   :grp1 
   Set Name     :set1
   Status       :Group-OutOfService
   Status (use) :Group-None

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 

Session information --
Session-id:35 
  Group:grp1  /*this session belongs to the group named 'grp1' */
Configuration:
     Local:10.1.2.15      , port:8303 
    Remote:10.5.0.3       , port:8303 
  Priority:2
  RUDP Option:Client, Conn Id:0x2
State:
  Status:OPEN_WAIT, Use-status:OOS,  /*see explanation below */
Statistics:
  # of resets:0
  # of auto_resets 0
  # 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
Reset:0 
  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
Reset:0 
  RUDP statistics 
         Open failures:0
         Not ready failures:0
         Conn Not Open failures:0
         Send window full  failures:0
         Resource unavailble failures:0
         Enqueue 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

GENERIC:
SetupTime=104443 ms
Index=1
PeerAddress=50110
PeerSubAddress=
PeerId=100
PeerIfIndex=104
LogicalIfIndex=10
ConnectTime=104964
CallDuration=00:02:43
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=2
TransmitPackets=15720
TransmitBytes=2362904
ReceivePackets=15670
ReceiveBytes=2737904
TELE:
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
TxDuration=155310 ms
VoiceTxDuration=155310 ms
FaxTxDuration=0 ms
CoderTypeRate=g711ulaw
NoiseLevel=-75
ACOMLevel=11
OutSignalLevel=-13
InSignalLevel=-22
InfoActivity=2
ERLLevel=27
SessionTarget=
ImgPages=0
 GENERIC:
SetupTime=104648 ms
Index=1
PeerAddress=55240
PeerSubAddress=
PeerId=2
PeerIfIndex=105
LogicalIfIndex=0
ConnectTime=104964
CallDuration=00:02:47
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=16026
TransmitBytes=2608248
ReceivePackets=16075
ReceiveBytes=2609164
VOIP:
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
RemoteUDPPort=18202
RoundTripDelay=2 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=TRUE

SessionProtocol=cisco
SessionTarget=ipv4:1.14.82.14
OnTimeRvPlayout=40
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
ReceiveDelay=67 ms
LostPackets=0 ms
EarlyPackets=0 ms
LatePackets=0 ms
VAD = enabled
CoderTypeRate=g729r8
CodecBytes=20
SignalingType=cas

Modem passthrough signaling method is nse:
Buffer Fill Events = 0
Buffer Drain Events = 0
Percent Packet Loss = 0
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 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

GENERIC:
SetupTime=22021 ms
Index=1
PeerAddress=wook song
PeerSubAddress=
PeerId=0
PeerIfIndex=0
LogicalIfIndex=0
ConnectTime=24284
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=10
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=41190

MMOIP:
ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]
RemoteIPAddress=0.0.0.0
SessionProtocol=SMTP
SessionTarget=
MessageId=
AccountId=
ImgEncodingType=MH
ImgResolution=fine
AcceptedMimeTypes=2
DiscardedMimeTypes=1
Notification=None

GENERIC:
SetupTime=23193 ms
Index=1
PeerAddress=527....
PeerSubAddress=
PeerId=3469
PeerIfIndex=157
LogicalIfIndex=30
ConnectTime=24284
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=10
TransmitPackets=5
TransmitBytes=6513
ReceivePackets=0
ReceiveBytes=0

TELE:     
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
TxDuration=24010 ms
FaxTxDuration=10910 ms
FaxRate=14400
NoiseLevel=-1
ACOMLevel=-1
OutSignalLevel=0
InSignalLevel=0
InfoActivity=0
ERLLevel=-1
SessionTarget=
ImgPages=0

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
tx:0/0 rx:0/41190
IP 0.0.0.0 AcceptedMime:2 DiscardedMime:1
 
1    : 23193hs.1 +1091 pid:3469 Originate 527.... active
tx:10/13838 rx:0/0
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
          and goto state start
    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
    Do Action IVR_ACT_END.
    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
          and do nothing
State get_account has 4 actions and 7 events
    Do Action IVR_ACT_PLAY.
            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
            patName=account
    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_PLAY.
            URL: flash:enter_pin.au
            allowInt=1, pContent=0x0
    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
            patName=pin
    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
    Do Action IVR_ACT_PLAY.
            URL: flash:enter_destination.au
            allowInt=1, pContent=0x0
    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.
            destination= called=
            calling=      account=
    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
    Do Action IVR_ACT_PLAY.
            URL: flash:auth_failed.au
            allowInt=0, pContent=0x0
    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
          and goto state start
    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
    Do Action IVR_ACT_END.
    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
          and do nothing
 State get_account has 4 actions and 7 events
    Do Action IVR_ACT_PLAY.
            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
            patName=account
    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_PLAY.
            URL: flash:enter_pin.au
            allowInt=1, pContent=0x0
    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
            patName=pin
    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
    Do Action IVR_ACT_PLAY.
            URL: flash:enter_destination.au
            allowInt=1, pContent=0x0
    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.
            destination= called=
            calling=      account=
    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
    Do Action IVR_ACT_PLAY.
            URL: flash:auth_failed.au
            allowInt=0, pContent=0x0
    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
 
2 active probes

Field                      Description
-------                    ------------
Probe                      Probe number
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

1 active probes

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

VoIP fallback config:
Fallback is ON
Using ICPIF threshold:
        ICPIF value timeout:20 seconds
        ICPIF threshold:20
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 
MD5 Keychain:secret

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

VOIP Fallback Stats:
Total accepted calls:3
Total rejected calls:1
Total cache overflows:1

Field                      Description
-------                    ------------
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

GENERIC:
SetupTime=104648 ms
Index=1
PeerAddress=55240
PeerSubAddress=
PeerId=2
PeerIfIndex=105
LogicalIfIndex=0
DisconnectCause=10  
DisconnectText=normal call clearing.
ConnectTime=104964
DisconectTime=143329
CallDuration=00:06:23
CallOrigin=1
ChargedUnits=0
InfoType=speech
TransmitPackets=37668
TransmitBytes=6157536
ReceivePackets=37717
ReceiveBytes=6158452
VOIP:
ConnectionId[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
RemoteIPAddress=1.14.82.14
RemoteUDPPort=18202
RoundTripDelay=2 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=TRUE

SessionProtocol=cisco
SessionTarget=ipv4:1.14.82.14
OnTimeRvPlayout=40
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=67 ms
LoWaterPlayoutDelay=67 ms
ReceiveDelay=67 ms
LostPackets=0 ms
EarlyPackets=0 ms
LatePackets=0 ms
VAD = enabled
CoderTypeRate=g729r8
CodecBytes=20
cvVoIPCallHistoryIcpif=0
SignalingType=cas

Modem passthrough signaling method is nse
Buffer Fill Events = 0
Buffer Drain Events = 0
Percent Packet Loss = 0
Consecutive-packets-lost Events = 0
Corrected packet-loss Events = 0
Last Buffer Drain/Fill Event = 373sec
Time between Buffer Drain/Fills = Min 0sec Max 0sec

GENERIC:
SetupTime=104443 ms
Index=2
PeerAddress=50110
PeerSubAddress=
PeerId=100
PeerIfIndex=104
LogicalIfIndex=10
DisconnectCause=10  
DisconnectText=normal call clearing.
ConnectTime=104964
DisconectTime=143330
CallDuration=00:06:23
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=37717
TransmitBytes=5706436
ReceivePackets=37668
ReceiveBytes=6609552
TELE:
ConnectionId=[0x4B091A27 0x3EDD0003 0x0 0xFEFD4]
TxDuration=375300 ms
VoiceTxDuration=375300 ms
FaxTxDuration=0 ms
CoderTypeRate=g711ulaw
NoiseLevel=-75
ACOMLevel=11
SessionTarget=
ImgPages=0

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

GENERIC:
SetupTime=23193 ms
Index=1
PeerAddress=527....
PeerSubAddress=
PeerId=3469
PeerIfIndex=157
LogicalIfIndex=30
DisconnectCause=10  
DisconnectText=normal call clearing.: Normal connection
ConnectTime=24284
DisconectTime=31288
CallOrigin=1
ChargedUnits=0
InfoType=fax
TransmitPackets=62
TransmitBytes=88047
ReceivePackets=0
ReceiveBytes=0

TELE:
ConnectionId=[0x37EC7F41 0xB0110001 0x0 0x35C34]
TxDuration=80950 ms
FaxTxDuration=10910 ms
FaxRate=14400
NoiseLevel=-1
ACOMLevel=-1
SessionTarget=
ImgPages=3

GENERIC:
SetupTime=22021 ms
Index=2
PeerAddress=wook song
PeerSubAddress=
PeerId=0
PeerIfIndex=0
LogicalIfIndex=0
DisconnectCause=10  
DisconnectText=normal call clearing.
ConnectTime=24284
DisconectTime=31545
CallOrigin=2
ChargedUnits=0
InfoType=fax
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=41190

MMOIP:
ConnectionId[0x37EC7F41 0xB0110001 0x0 0x35C34]
RemoteIPAddress=0.0.0.0
SessionProtocol=SMTP
SessionTarget=
MessageId=
AccountId=
ImgEncodingType=MH
ImgResolution=fine
AcceptedMimeTypes=2
DiscardedMimeTypes=1
Notification=None

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

CallId = 4
CalledNumber = 221
CallDuration = 39006 seconds
DisconnectText = remote hangup
SVC: call ID = 8598630
Remote NSAP = 47.0091810000000002F26D4901.00107B09C645.C8
Local NSAP = 47.0091810000000002F26D4901.00107B4832E1.C8
vcd = 414, vpi = 0, vci = 158
SerialPort = Serial0
VideoSlot = 1, VideoPort = 0
CallId = 3
CalledNumber = 221
CallDuration = 557 seconds
DisconnectText = local hangup
SVC: call ID = 8598581
Remote NSAP = 47.0091810000000002F26D4901.00107B09C645.C8
Local NSAP = 47.0091810000000002F26D4901.00107B4832E1.C8
vcd = 364, vpi = 0, vci = 108
SerialPort = Serial0
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
CallingNumber=2001
SetupTime=1157801 x 10ms
ConnectTime=1158046 x 10ms
DisconectTime=1158188 x 10ms
DisconnectText=local onhook
 
ConnectionId=[0x2C7AEFDC 0x59830001 0x0 0xB0AAA3]
Media=TELE, TxDuration= 1422 ms
CalledNumber=2002
SetupTime=1157802 x 10ms
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:

DSP Statistics:

Utilization: 0 percent
Total channels: 48
Inuse channels: 0
Disabled channels 0:
Pending channels: 0
Free channels: 48

DS0 Statistics:

Total channels: 0
Addressable channels: 0
Inuse channels: 0
Disabled channels: 0
Free channels: 0

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:

DS0 Threshold:

Client Type: h323
High Water Mark: 70
Low Water Mark: 60
Threshold State: init
DSP Threshold:

Client Type: h323
High Water Mark: 70
Low Water Mark: 60
Threshold State: low_threshold_hit

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:

Router# show cdapi

Registered CDAPI Applications/Stacks
====================================
Application TSP CDAPI Application
        Application Type(s)  Voice Facility Signaling 
        Application Level    Tunnel
        Application Mode     Enbloc
Signaling Stack ISDN
        Interface Se023
Signaling Stack ISDN
        Interface Se123
Active CDAPI Calls
==================
Interface Se023
        No active calls.
Interface Se123
        Call ID = 0x39, Call Type = VOICE, Application = TSP CDAPI Application
CDAPI Message Buffers
=====================
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
scarlatti1# 

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:

Router# show connect all

ID   Name               Segment 1            Segment 2           State
========================================================================
1    Test              -T1 1/0 01           -T1 1/1 02           ADMIN UP
2    Test2             -T1 1/0 03           -T1 1/1 04           ADMIN UP

Router# show connect id 1-2

ID   Name               Segment 1            Segment 2           State
========================================================================
1    Test              -T1 1/0 01           -T1 1/1 02           ADMIN UP
2    Test2             -T1 1/0 03           -T1 1/1 04           ADMIN UP

Router# show connect port t1 1/1

ID   Name               Segment 1            Segment 2           State
========================================================================
1    Test              -T1 1/0 01           -T1 1/1 02           ADMIN UP
2    Test2             -T1 1/0 03           -T1 1/1 04           ADMIN UP

The following examples show details about specific connections, including the number of time slots in use and the switching elements:

Router# show connect id 2

Connection: 2 - Test2
 Current State: ADMIN UP
 Segment 1: -T1 1/0 03
  TDM timeslots in use: 14-18 (5 total)
 Segment 2: -T1 1/1 04
  TDM timeslots in use: 14-18
Internal Switching Elements: VIC TDM Switch

Router# show connect name Test

 Connection: 1 - Test
 Current State: ADMIN UP
 Segment 1: -T1 1/0 01
  TDM timeslots in use: 1-13 (13 total)
 Segment 2: -T1 1/1 02
  TDM timeslots in use: 1-13
Internal Switching Elements: VIC TDM Switch

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  
Dial string:
121C

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
T1 1 is up:
Loopback: NONE
DS0  Type       Modem    <->     Service       Channel       Rx          Tx
                                 State         State        A B C D      A B C D
-----------------------------------------------------------------------------------------
  1    cas-modem   1       in     insvc       connected    1  1  1  1    1  1  1  1 
  2    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0 
  3    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  4    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  5    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  6    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  7    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  8    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  9    cas         -       -      insvc       idle         0  0  0  0    0  0  0  0
  10   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  11   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  12   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  13   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  14   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  15   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  16   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  17   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  18   cas         -       -      maint      static-bo     0  0  0  0    1  1  1  1 
  19   cas         -       -      maint      dynamic-bo    0  0  0  0    1  1  1  1 
  20   cas         -       -      maint      dynamic-bo    0  0  0  0    1  1  1  1 
  21   cas         -       -      maint      dynamic-bo    0  0  0  0    1  1  1  1 
  22   unused
  23   unused
  24   unused

The following example shows that the ISDN PRI state is enabled on the Cisco AS5300 universal access server with a T1 PRI card:

T1 2 is up:
Loopback: NONE
DS0 Type         Modem    <->  Service   Channel     Rx        Tx
                               State     State       A B C D   A B C D
---------------------------------------------------------------------------
 1  pri          -        -    insvc     idle       
 2  pri          -        -    insvc     idle       
 3  pri          -        -    insvc     idle       
 4  pri          -        -    insvc     idle       
 5  pri          -        -    insvc     idle       
 6  pri          -        -    insvc     idle       
 7  pri          -        -    insvc     idle       
 8  pri          -        -    insvc     idle       
 9  pri          -        -    insvc     idle       
10  pri          -        -    insvc     idle       
11  pri          -        -    insvc     idle       
12  pri          -        -    insvc     idle       
13  pri          -        -    insvc     idle       
14  pri          -        -    insvc     idle       
15  pri          -        -    insvc     idle       
16  pri          -        -    insvc     idle       
17  pri          -        -    insvc     idle       
18  pri          -        -    insvc     idle       
19  pri          -        -    insvc     idle       
20  pri          -        -    insvc     idle       
21  pri-modem    2        in   insvc     busy       
22  pri-modem    1        out  insvc     busy       
23  pri-digi     -        in   insvc     busy       
24  pri-sig      -        -    outofsvc  reserved 

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

EPIC Switch registers:
STDA 0xFF STDB 0xFF SARA 0xAD SARB 0xFF SAXA 0xFF SAXB 0x0 STCR 0x3F
MFAIR 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
0x0

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

Rx Message 0:
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

Rx Message 1:
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

Rx Message 2:
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

Rx Message 3:
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

Rx Message 4:
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

Rx Message 5:
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

Tx Message 0:
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

Tx Message 1:
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

Tx Message 2:
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

Tx Message 3:
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

Tx Message 4:
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

Tx Message 5:
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

Tx Message 6:
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

Tx Message 7:
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

Tx Message 8:
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

Bootloader 1.8, Appn 3.1
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
underrun
0 unaligned frames

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
underrun
0 unaligned frames

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
oobp_failure=0
call_duration_started=00:06:53, call_duration_ended=00:00:00, total_call_duration=00:00:44
The calling party phone number = 408
The called party phone number  = 5271086
total_free_rbs_timeslot = 0, total_busy_rbs_timeslot = 0, total_dynamic_busy_rbs_timeslot 
= 0, total_static_busy_rbs_timeslot = 0,
total_sw56_rbs_timeslot = 0, total_sw56_rbs_static_bo_ts = 0,
total_free_isdn_channels = 21, total_busy_isdn_channels = 0,total_auto_busy_isdn_channels 
= 0, 
min_free_device_threshold = 0

The following is sample output from the Cisco AS5800 for the show csm voice command:

Router# show csm voice 1/8/19

 shelf 1, slot 8, port 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
call thru PRI line
invalid_event_count=0, wdt_timeout_count=0
watchdog timer is not activated
wait_for_bchan:False
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

Video Dial-Peer 1
      type = videocodec, destination-pattern = 111
      port signal = 1/0, port media = Serial1
      nsap = 47.0091810000000050E201B101.00107B09C6F2.C8
Video Dial-Peer 2
      type = videoatm,   destination-pattern = 222
      session-target = ATM0 svc nsap 47.0091810000000050E201B101.00E01E92ADC2.C8
Video Dial-Peer 3
      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

VoiceEncapPeer1
        tag = 1, dest-pat = `+14085291000',
        answer-address = `',
        group = 0, Admin state is up, Operation state is down
        Permission is Both,
        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 ""
        Last Setup Time = 0

The following is sample output from the show dial-peer voice command for a VoIP dial peer:

Router# show dial-peer voice 10

VoiceOverIpPeer10
        tag = 10, dest-pat = `',
        incall-number = `+14087',
        group = 0, Admin state is up, Operation state is down
        Permission is Answer, 
        type = voip, session-target = `',
        sess-proto = cisco, req-qos = bestEffort, 
        acc-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 ""
        Last Setup Time = 0

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

Macro Exp.: 12345

VoiceEncapPeer10
        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,
        DTMF Relay = disabled,
        huntstop = disabled,
        in bound application associated: DEFAULT
        out bound application associated: 
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        type = pots, prefix = `',
        forward-digits default
        session-target = `', voice-port = `1/0/0:D',
        direct-inward-dial = disabled,
        digit_strip = enabled,

        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 "",
        Last Setup Time = 0.
Matched: 12345   Digits: 3
Target: 

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

Macro Exp.: 888

VoiceEncapPeer99
        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,
        DTMF Relay = disabled,
        huntstop = disabled,
        in bound application associated: DEFAULT
        out bound application associated: 
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        type = pots, prefix = `5',
        forward-digits 4
        session-target = `', voice-port = `1/0/1:D',
        direct-inward-dial = enabled,
        digit_strip = 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 "",
        Last Setup Time = 0.
Matched:    Digits: 0
Target: 

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 

Macro Exp.: 1001

VoiceEncapPeer1003
         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,
         DTMF Relay = disabled,
         huntstop = enabled,
         type = pots, prefix = `',
         forward-digits default
         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 "",
         Last Setup Time = 0.
Matched: 1001   Digits: 4
Target: 

VoiceEncapPeer1004
         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,
...
Matched: 1001   Digits: 4
Target: 

VoiceEncapPeer1002
         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,
...
Matched: 1001   Digits: 4
Target: 

VoiceEncapPeer1001
         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,
...
Matched: 1001   Digits: 4
Target: 

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

 Macro Exp.: 1001

 VoiceEncapPeer1003
         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,
         DTMF Relay = disabled,
         huntstop = enabled,
         type = pots, prefix = `',
         forward-digits default
         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 "",
         Last Setup Time = 0.
 Matched: 1001   Digits: 4
 Target: 

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     4     data
Serial0           VoFR        16     5     call-control
Serial0           VoFR        16     10    voice

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       4   data              0             0             0
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.
                         GATEKEEPER CALL INFO
                         ====================
LocalCallID                        Age(secs)   BW
12-3339                            94          768(Kbps)
 Endpt(s):Alias        E.164Addr     CallSignalAddr  Port  RASSignalAddr   Port
   src EP:epA                        90.0.0.11       1720  90.0.0.11       1700
   dst EP:epB@zoneB.com                      
   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:
    172.21.13.11:1720 sj-gw1
    172.21.13.22:1720 sj-gw2 (out-of-resources)
    172.21.13.33:1720 sj-gw3
  Zone sj-gk prefix 408....... priority gateway list(s):
   Priority 10:
    172.21.13.11:1720 sj-gw1
   Priority 5:
    172.21.13.22:1720 sj-gw2 (out-of-resources)
    172.21.13.33:1720 sj-gw3
Prefix:7#*     (Hopoff zone la-gk)
  Statically-configured gateways (not necessarily currently registered):
    1.1.1.1:1720
    2.2.2.2:1720
  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

     Gatekeeper-ID:gk102
     --------------------
     RRQ  Priority:1
       Server-ID:sj-server
       Server IP address:1.14.93.28:42387
       Server type:dynamically registered
       Connection Status:active
       Trigger Information:
         Supported Prefix:10#
         Supported Prefix:3#
     RRQ  Priority:2
       Server-ID:sf-server
       Server IP address:1.14.93.43:3820
       Server type:CLI-configured
       Connection Status:inactive
       Trigger Information:
         Endpoint-type:MCU
         Endpoint-type:VOIP-GW
         Supported Prefix:99#
     ARQ  Priority:1
       Server-ID:sj-server
       Server IP address:1.14.93.28:42387
       Server type:dynamically registered
       Connection Status:active
       Trigger Information:
         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

Gatekeeper State: UP
Zone Name: gk-px4.cisco.com
Accounting: DISABLED
Security: DISABLED

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

      ZONE PREFIX TABLE
      =================
GK-NAME               E164-PREFIX
-------               -----------
gk.zone13             212.......
gk.zone14             415.......
gk.zone14             408.......

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

                      GATEKEEPER ZONES
                      ================
GK name      Domain Name   RAS Address     PORT  FLAGS MAX-BW   CUR-BW
                                                       (kbps)   (kbps)
-------      -----------   -----------     ----  ----- ------   ------
sj.xyz.com   xyz.com       1.14.93.85      1719  LS             0       
  SUBNET ATTRIBUTES :
    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:

Router# show gateway

Gateway gateway1 is not registered to any gatekeeper
Gateway alias list 
H323-ID gateway1
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:

Router# show gate

Gateway  gateway1 is registered to Gatekeeper gk1
Gateway alias list
E.164 Number 5551212
H323-ID gateway1

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:

Router# show gateway

Gateway gateway1 is registered to Gatekeeper gk1
Gateway alias list 
H323-ID gateway1
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:

Router# show gateway

Gateway gateway1 is registered to Gatekeeper gk1
Gateway alias list 
H323-ID gateway1
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
  Hardware is VXC-2T1/E1
  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)
     DSP Application:4.0 (3)
     Medium Complexity Application:3.2 (5)
     High Complexity Application:3.2 (5)
  Total DSPs 30, DSP0-DSP29, Jukebox DSP id 30
  Down DSPs:none
  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
  Queueing strategy:fifo
  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.

Router# 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 modem passthru: CA
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

UDP pkts rx 8, tx 9
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
ACK tx 8, NACK tx 0
ACK rx 0, NACK rx 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:

Router# show num-exp

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

port

Port number 1 or 2.


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:

Router# show pots csm 1

POTS PORT: 1

   CSM Finite State Machine:
      Call 0 - State: idle, Call Id: 0x0
               Active: no
               Event: CSM_EVENT_NONE Cause: 0
      Call 1 - State: idle, Call Id: 0x0
               Active: no
               Event: CSM_EVENT_NONE Cause: 0
      Call 2 - State: idle, Call Id: 0x0
               Active: no
               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.

Router# show pots status

POTS Global Configuration:
   Country: United States
   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,
   Filter Mask: 6F
   Adaptive Cntrl Mask: 0
POTS PORT: 1
   Hook Switch Finite State Machine:
      State: On Hook, Event: 0
      Hook Switch Register: 10, Suspend Poll: 0
   CODEC Finite State Machine:
      State: Idle, Event: 0
      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,
      Filter Mask: 6F
      Adaptive Cntrl Mask: 0
   CODEC Registers:
      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
      SLIC Pin Direction: 1F
   CODEC Coefficients:
      GX: A0 00
      GR: 3A A1
       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
      GZ: 60
     ADAPT B: 91 B2 8F 62 31
   CSM Finite State Machine:
      Call 0 - State: idle, Call Id: 0x0
               Active: no
      Call 1 - State: idle, Call Id: 0x0
               Active: no
      Call 2 - State: idle, Call Id: 0x0
               Active: no
POTS PORT: 2
   Hook Switch Finite State Machine:
      State: On Hook, Event: 0
      Hook Switch Register: 20, Suspend Poll: 0
   CODEC Finite State Machine:
      State: Idle, Event: 0
      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,
      Filter Mask: 6F
      Adaptive Cntrl Mask: 0
   CODEC Registers:
      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
      SLIC Pin Direction: 1F
   CODEC Coefficients:
      GX: A0 00
      GR: 3A A1
       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
      GZ: 60
     ADAPT B: 91 B2 8F 62 31
   CSM Finite State Machine:
      Call 0 - State: idle, Call Id: 0x0
               Active: no
      Call 1 - State: idle, Call Id: 0x0
               Active: no
      Call 2 - State: idle, Call Id: 0x0
               Active: no
Time Slot Control: 0

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

Call unique key = 1
  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
Called endpoint aliases:
      H323_ID: ptel21@zone2.com
Peer proxy call signalling address = 55.0.0.41
Time call was initiated = 731146290 ms
Inbound CRV = 144
Outbound CRV = 70
Call state = Media Streaming
H245 logical channels for call leg pte111@zone1.com<->px1@zone.com
    Channel number = 2
        Type = VIDEO
        State = OPEN
        Bandwidth = 374 kbps
        Time created = 731146317 ms
    Channel number = 1
        Type = AUDIO
        State = OPEN
        Bandwidth = 81 kbps
        Time created = 731146316 ms
    Channel number = 2
        Type = VIDEO
        State = OPEN
        Bandwidth = 374 kbps
        Time created = 731146318 ms
    Channel number = 1
        Type = AUDIO
        State = OPEN
        Bandwidth = 81 kbps
        Time created = 731146317 ms
H245 logical channels for call leg pte111@zone1.com<->50.0.0.41:
    Channel number = 2
        Type = VIDEO
        State = OPEN
        Bandwidth = 374 kbps
        Time created = 731146317 ms
    Channel number = 1
        Type = AUDIO
        State = OPEN
        Bandwidth = 81 kbps
        Time created = 731146316 ms
    Channel number = 2
        Type = VIDEO
        State = OPEN
        Bandwidth = 374 kbps
        Time created = 731146318 ms
    Channel number = 1
        Type = AUDIO
        State = OPEN
        Bandwidth = 81 kbps
        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: intel2
      E164_ID: 2134
Called endpoint aliases: