Table Of Contents
Cisco IOS Voice Commands:
V
vad (dial peer)
vbd-playout-delay maximum
vbd-playout-delay minimum
vbd-playout-delay mode
vbd-playout-delay nominal
vbr-rt
vcci
video codec (dial peer)
video codec (voice class)
vofr
voice
voice call capacity mir
voice call capacity reporting
voice call capacity stw
voice call capacity timer interval
voice call convert-discpi-to-prog
voice call csr data-points
voice call csr recording interval
voice call csr reporting interval
voice call debug
voice call disc-pi-off
voice call send-alert
voice call trigger hwm
voice call trigger lwm
voice call trigger percent-change
voicecap configure
voicecap entry
voice-card
voice class aaa
voice-class aaa (dial peer)
voice class busyout
voice-class called-number (dial peer)
voice class called number
voice-class called-number-pool
voice class codec
voice-class codec (dial peer)
voice class custom-cptone
voice class dualtone
voice class dualtone-detect-params
Cisco IOS Voice Commands:
V
This chapter contains commands to configure and maintain Cisco IOS voice applications. The commands are presented in alphabetical order. Some commands required for configuring voice 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 Configuration Guide.
vad (dial peer)
To enable voice activity detection (VAD) for the calls using a particular dial peer, use the vad command in dial peer configuration mode. To disable VAD, use the no form of this command.
vad [aggressive]
no vad [aggressive]
Syntax Description
aggressive
|
Reduces noise threshold from -78 to -62 dBm. Available only when session protocol multicast is configured.
|
Command Default
VAD is enabled
Aggressive VAD is enabled in multicast dial peers
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on Cisco 3600 series.
|
12.0(4)T
|
This command was implemented as a dial-peer command on Cisco MC3810 (in prior releases, the vad command was available only as a voice-port command).
|
12.2(11)T
|
The aggressive keyword was added.
|
Usage Guidelines
Use this command to enable voice activity detection. With VAD, voice data packets fall into three categories: speech, silence, and unknown. Speech and unknown packets are sent over the network; silence packets are discarded. The sound quality is slightly degraded with VAD, but the connection monopolizes much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously sent to the IP backbone. When configuring voice gateways to handle fax calls, VAD should be disabled at both ends of the IP network because it can interfere with the successful reception of fax traffic.
When the aggressive keyword is used, the VAD noise threshold is reduced from -78 to -62 dBm. Noise that falls below the -62 dBm threshold is considered to be silence and is not sent over the network. Additionally, unknown packets are considered to be silence and are discarded.
Examples
The following example enables VAD for a Voice over IP (VoIP) dial peer, starting from global configuration mode:
Related Commands
Command
|
Description
|
comfort-noise
|
Generates background noise to fill silent gaps during calls if VAD is activated.
|
dial-peer voice
|
Enters dial peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
|
vad (voice-port)
|
Enables VAD for the calls using a particular voice port.
|
vbd-playout-delay maximum
To enable maximum ATM adaptation layer 2 (AAL2) voice-band-detection playout-delay buffer on a Cisco router, use the vbd-playout-delay command in voice-service configuration mode. To reset to the default, use the no form of this command.
vbd-playout-delay maximum time
no vbd-playout-delay maximum
Syntax Description
time
|
Playout delay, in milliseconds. Range is from 40 to 1700. The default is 200.
|
Command Default
200 milliseconds
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on Cisco 2600 series and Cisco 3660 routers.
|
Examples
The following example sets the AAL2 voice-band-detection playout-buffer delay to a maximum of 202 milliseconds:
voice service voatm
session protocol aal2
vbd-playout-delay maximum 202
Related Commands
Command
|
Description
|
voice-service
|
Specifies the voice encapsulation type and enters voice-service configuration mode.
|
vbd-playout-delay minimum
To enable minimum ATM adaptation layer 2 (AAL2) voice-band-detection playout-delay buffer on a Cisco router, use the vbd-playout-delay minimum command in voice-service configuration mode. To reset to the default, use the no form of this command.
vbd-playout-delay minimum time
no vbd-playout-delay minimum
Syntax Description
time
|
Playout delay, in milliseconds. Range is from 4 to 1700. The default is 4.
|
Command Default
4 milliseconds
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on Cisco 2600 series and Cisco 3660 routers.
|
Examples
The following example sets the AAL2 voice-band-detection playout-buffer delay to a minimum of 6 milliseconds:
voice service voatm
session protocol aal2
vbd-playout-delay minimum 6
Related Commands
Command
|
Description
|
voice-service
|
Specifies the voice encapsulation type and enters voice-service configuration mode.
|
vbd-playout-delay mode
To configure voice-band-detection playout-delay adaptation mode on a Cisco router, use the vbd-playout-delay mode command in voice-service configuration mode. To disable this mode, use the no form of this command.
vbd-playout-delay mode [fixed | passthrough]
no vbd-playout-delay mode [fixed | passthrough]
Syntax Description
fixed
|
Sets jitter buffer to a constant delay, in milliseconds.
|
passthrough
|
Sets jitter buffer passthrough to DRAIN_FILL for clock compensation.
|
Command Default
Voice-band-detection playout-delay adaptation mode is disabled.
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on Cisco 2600 series and Cisco 3660 routers.
|
Usage Guidelines
Use this command to set the playout jitter buffer. When a voice band is detected, the call uses G.711 codec, and the playout delay values that you set are picked up. The original voice-call parameters are restored after the fax or modem call is completed.
Examples
The following example configures ATM adaptation layer 2 (AAL2) voice-band-detection playout-delay adaptation mode and sets the mode to fixed:
vbd-playout-delay mode fixed
Related Commands
Command
|
Description
|
voice-service
|
Specifies the voice encapsulation type and enters voice-service configuration mode.
|
vbd-playout-delay nominal
To enable nominal ATM adaptation layer 2 (AAL2) voice-band-detection playout-delay buffer on a Cisco router, use the vbd-playout-delay command in voice-service configuration mode. To reset to the default, use the no form of this command.
vbd-playout-delay nominal time
no vbd-playout-delay nominal
Syntax Description
time
|
Playout delay, in milliseconds. Range is from 0 to 1500. The default is 100.
|
Command Default
100 milliseconds
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on Cisco 2600 series and Cisco 3660 routers.
|
Examples
The following example sets the nominal AAL2 voice-band-detection playout-delay buffer to 202 milliseconds:
voice service voatm
session protocol aal2
vbd-playout-delay nominal 202
Related Commands
Command
|
Description
|
voice-service
|
Specifies the voice encapsulation type and enters voice-service configuration mode.
|
vbr-rt
To configure the real-time variable bit rate (VBR) for VoATM voice connections, use the vbr-rt command in the appropriate configuration mode. To disable VBR for voice connections, use the no form of this command.
vbr-rt peak-rate average-rate burst
no vbr-rt
Syntax Description
peak-rate
|
Peak information rate (PIR) for the voice connection, in kbps. If it does not exceed your carrier's line rate, set it to the line rate. Range is from 56 to 10000.
|
average-rate
|
Average information rate (AIR) for the voice connection in kbps.
|
burst
|
Burst size, in number of cells. Range is from 0 to 65536.
|
Command Default
No real-time VBR settings are configured
Command Modes
For an ATM permanent virtual connection (PVC) or switched virtual circuit (SVC): Interface-ATM-VC configuration
For a virtual circuit (VC) class: VC-class configuration
For ATM VC bundle members: Bundle-vc configuration
Command History
Release
|
Modification
|
12.0
|
This command was introduced on Cisco MC3810.
|
12.1(5)XM
|
This command was implemented on Cisco 3600 series routers and modified to support Simple Gateway Control Protocol (SGCP) and Media Gateway Control Protocol (MGCP).
|
12.2(2)T
|
This command was integrated into Cisco IOS Release 12.2(2)T.
|
12.2(11)T
|
This command was implemented on the Cisco AS5300 and Cisco AS5850.
|
Usage Guidelines
This command configures traffic shaping between voice and data PVCs. Traffic shaping is required so that the carrier does not discard calls. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will carry bursty traffic. Peak, average, and burst values are needed so that the PVC can effectively handle the bandwidth for the number of voice calls.
Calculate the minimum peak, average, and burst values for the number of voice calls as follows:
Peak Value
Peak value = (2 x the maximum number of calls) x 16K = _______________
Average Value
Calculate according to the maximum number of calls that the PVC will carry times the bandwidth per call. The following formulas give you the average rate in kbps:
•For VoIP:
–G.711 with 40- or 80-byte sample size:
Average value = max calls x 128K = _______________
–G.726 with 40-byte sample size:
Average value = max calls x 85K = _______________
–G.729a with 10-byte sample size:
Average value = max calls x 85K = _______________
•For VoATM adaptation layer 2 (VoAAL2):
–G.711 with 40-byte sample size:
Average value = max calls x 85K = _______________
–G.726 with 40-byte sample size:
Average value = max calls x 43K = _______________
–G.729a with 10-byte sample size:
Average value = max calls x 43K = _______________
If voice activity detection (VAD) is enabled, bandwidth usage is reduced by as much as 12 percent with the maximum number of calls in progress. With fewer calls in progress, bandwidth savings are less.
Burst Value
Set the burst size as large as possible, and never less than the minimum burst size. Guidelines are as follows:
•Minimum burst size = 4 x number of voice calls = _______________
•Maximum burst size = maximum allowed by the carrier = _______________
When you configure data PVCs that will be traffic shaped with voice PVCs, use aal5snap encapsulation and calculate the overhead as 1.13 times the voice rate.
Examples
The following example configures the traffic-shaping rate for ATM PVC 20. Peak, average, and burst rates are calculated based on a maximum of 20 calls on the PVC.
encapsulation aal5mux voice
Related Commands
Command
|
Description
|
encapsulation aal5
|
Configures the AAL and encapsulation type for an ATM PVC, SVC, or VC class.
|
vcci
To identify a permanent virtual circuit (PVC) to the call agent, use the vcci command in ATM virtual circuit (VC) configuration mode. To restore the default value, use the no form of this command.
vcci pvc-identifier
no vcci
Syntax Description
pvc-identifier
|
Identifier for the PVC. Range is from 0 to 32767. There is no default value.
|
Command Default
No default behavior or values
Command Modes
ATM virtual circuit configuration mode
Command History
Release
|
Modification
|
12.1(5)XM
|
This command was introduced.
|
12.2(2)T
|
This command was integrated into Cisco IOS Release 12.2(2)T.
|
12.2(11)T
|
This command was implemented on the Cisco AS5300 and Cisco AS5850.
|
Usage Guidelines
The pvc-identifier argument is a unique 15-bit value for each PVC. The call agent sets up a call with the gateway by specifying the PVC using the pvc-identifier.
Examples
The following example shows how to assign a PVC identifier:
Router(config-if-atm-vc)# vcci 5278
Related Commands
Command
|
Description
|
mgcp
|
Starts the MGCP daemon.
|
pvc
|
Creates an ATM PVC for voice traffic.
|
video codec (dial peer)
To assign a video codec to a VoIP dial peer, use the video codec command in dial peer configuration mode. To remove a video codec, use the no form of this command.
video codec {h261 | h263 | h263+ | h264}
no video codec
Syntax Description
h261
|
Video codec H.261
|
h263
|
Video codec H.263
|
h263+
|
Video codec H.263+
|
h264
|
Video codec H.264
|
Command Default
No video codec is configured.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.4(11)T
|
This command was introduced.
|
Usage Guidelines
Use this command to configure a video codec for a VoIP dial peer. If no video codec is configured, the default is transparent codec operation between the endpoints.
Examples
The following example shows configuration for video codec H.263+ on VoIP dial peer 30:
Related Commands
Command
|
Description
|
video codec (voice-class)
|
Specifies a video codec for a voice class.
|
video codec (voice class)
To specify a video codec for a voice class, use the video codec command in voice class configuration mode. To remove the video codec, use the no form of this command.
video codec {h261 | h263 | h263+ | h264}
no video codec {h261 | h263 | h263+ | h264}
Syntax Description
h261
|
Apply this preference to video codec H.261
|
h263
|
Apply this preference to video codec H.263
|
h263+
|
Apply this preference to video codec H.263+
|
h264
|
Apply this preference to video codec H.264
|
Command Default
No video codec is configured.
Command Modes
Voice class configuration
Command History
Release
|
Modification
|
12.4(11)T
|
This command was introduced.
|
Usage Guidelines
Use this command to specify one or more video codecs for a voice class.
Examples
The following example shows configuration for voice class codec 10 with two audio codec preferences and three video codec preferences:
codec preference 1 g711alaw
Related Commands
Command
|
Description
|
video codec (dial peer)
|
Specifies a video codec for a VoIP dial peer.
|
vofr
To enable Voice over Frame Relay (VoFR) on a specific data-link connection identifier (DLCI) and to configure specific subchannels on that DLCI, use the vofr command in frame relay DLCI configuration mode. To disable VoFR on a specific DLCI, use the no form of this command.
Switched Calls
vofr [data cid] [call-control [cid]]
no vofr [data cid] [call-control [cid]]
Switched Calls to Cisco MC3810 Multiservice Concentrators Running Cisco IOS Releases Release Before 12.0(7)XK and Release 12.1(2)T
vofr [cisco]
no vofr [cisco]
Cisco-Trunk Permanent Calls
vofr data cid call-control cid
no vofr data cid call-control cid
FRF.11 Trunk Calls
vofr [data cid] [call-control cid]
no vofr [data cid] [call-control cid]
Syntax Description
data
|
(Required for Cisco-trunk permanent calls. Optional for switched calls.) Selects a subchannel (CID) for data other than the default subchannel, which is 4.
|
cid
|
(Optional) Specifies the subchannel to be used for data. Range is from 4 to 255. The default is 4. If data is specified, enter a valid CID.
|
call-control
|
(Optional) Reserves a subchannel for call-control signaling.
|
cisco
|
(Optional) Cisco proprietary voice encapsulation for VoFR with data is carried on CID 4 and call-control on CID 5.
|
cid
|
(Optional) Specifies the subchannel to be used for call-control signaling. Valid range is from 4 to 255. The default is 5. If call-control is specified and a CID is not entered, the default CID is used.
|
Command Default
Disabled
Command Modes
Frame relay DLCI configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and Cisco MC3810.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.0(7)XK
|
The use of the cisco option was modified. Beginning in this release, use the cisco option only when configuring connections to Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
Usage Guidelines
Table 239 lists the different options of the vofr command and which combination of options is used beginning in Cisco IOS Release 12.0(7)XK and Release 12.1(2)T.
Table 239 Combinations of the vofr Command
Type of Call
|
Command Combination to Use
|
Switched call (user dialed or auto-ringdown) to other routers supporting VoFR
|
vofr [data cid] [call-control [cid]]1
|
Cisco-trunk permanent call (private-line) to other routers supporting VoFR
|
vofr data cid call-control cid
|
FRF.11 trunk call (private-line) to other routers supporting VoFR
|
vofr [data cid] [call-control cid]2
|
Examples
The following example, beginning in global configuration mode, shows how to enable VoFR on serial interface 1/1, DLCI 100. The example configures CID 4 for data; no call-control CID is defined.
frame-relay interface-dlci 100
To configure CID 4 for data and CID 5 for call-control (both defaults), enter the following command:
To configure CID 10 for data and CID 15 for call-control, enter the following command:
vofr data 10 call-control 15
To configure CID 4 for data and CID 15 for call-control, enter the following command:
To configure CID 10 for data and CID 5 for call-control, enter the following command:
vofr data 10 call-control
To configure CID 10 for data with no call-control, enter the following command:
Related Commands
Command
|
Description
|
class
|
Assigns a VC class to a PVC.
|
frame-relay interface-dlci
|
Assigns a DLCI to a specified Frame Relay subinterface.
|
voice
To enable voice resource pool services for resource pool management, use the voice command in service profile configuration mode. To disable voice services, use the no form of this command.
voice
no voice
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Service profile configuration mode
Command History
Release
|
Modification
|
12.2(2)XA
|
This command was introduced on the Cisco AS5350 and AS5400.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850 platform.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T.
|
Examples
The following example shows that voice service is available and enables voice resource pool service using the voice command in service profile configuration mode:
Router(config)# resource-pool profile service voip
Router(config-service-profile)# ?
Service Profile Configuration Commands:
default Set a command to its defaults
exit Exit from resource-manager configuration mode
help Description of the interactive help system
modem Configure modem service parameters
no Negate a command or set in its defaults
voice Configure voice service parameters
Router(config-service-profile)# voice
Related Commands
Command
|
Description
|
resource-pool enable
|
Enables resource pool management.
|
resource-pool profile service voip
|
Defines the VoIP service profile for resource pool management.
|
voice call capacity mir
To set the value for the minimum interval between reporting (MIR), use the voice call capacity mir command in global configuration mode. To turn off these attributes, use the no form of this command.
voice call {carrier | trunk-group | prefix} capacity mir seconds
no voice call {carrier | trunk-group | prefix} capacity mir
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
seconds
|
Minimum interval, in seconds, with a range of 1 to 3600 seconds and a default of 10. This value cannot be set higher than the time configured for the capacity update interval.
|
Command Default
10 seconds.
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
Because the available circuit (AC) attribute of a destination is very dynamic, reporting of this attribute should be handled carefully. AC should be reported as frequently as possible so that the location server has better information about the resources. However, the location server should not be overwhelmed with too many updates.
All of the AC reporting, called the interesting point of AC, is performed when the specified event happens within the minimum interval between reporting (MIR) time since last reporting. This command sets the amount of time used for the interval to control the number of interesting points that are reported so not to overwhelm the location server with too many AC updates.
The seconds argument cannot be set higher than the time configured for the capacity update interval.
Examples
The following example shows the minimum interval between reporting for the carrier address family set to 25 seconds:
Router(config)# voice call carrier capacity mir 25
Related Commands
Command
|
Description
|
capacity update interval (dial peer)
|
Changes the capacity update for prefixes associated with a dial peer.
|
capacity update interval (trunk group)
|
Change the capacity update for carriers or trunk groups.
|
voice call capacity stw
|
Set the value for STW.
|
voice call capacity reporting
To turn on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity, use the voice call capacity reporting command in global configuration mode. To turn off the reporting, use the no form of this command.
voice call {carrier | trunk-group | prefix} capacity reporting {maxima | inflection}
no voice call {carrier | trunk-group | prefix} capacity reporting {maxima | inflection}
Syntax Description
carrier
|
Carrier code address family.
|
trunk-group
|
Trunk group address family.
|
prefix
|
E.164 prefix.
|
maxima
|
Maxima (first derivative) point in available capacity.
|
inflection
|
Inflection (second derivative) point in available capacity.
|
Command Default
The capacity reporting function is turned off.
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
The smoothed curve of the available circuits (AC) has maxima, minima, and inflection points. When the curve has reached these points, this represents a change in the call rate.
Maximum, minimum and inflection points are illustrated in Figure 5.
Figure 5 Maximum, Minimum, and Inflection Points for Available Capacity
Examples
The following example shows the reporting of the available capacity inflection point on the trunk group is turned on:
Router(config)# voice call trunk-group capacity reporting inflection
Related Commands
Command
|
Description
|
voice call capacity mir
|
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
|
voice call capacity timer interval
|
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases
|
voice call trigger hwm
|
Sets the value for percentage change, low water mark and high water mark in the available capacity in the trunk group or prefix databases.
|
voice call capacity stw
To set the value for smoothing transition time for weight (STW), use the voice call capacity stw command in global configuration mode. To turn off these attributes, use the no form of this command.
voice call {carrier | trunk-group | prefix} capacity stw seconds
no voice call {carrier | trunk-group | prefix} capacity stw seconds
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
seconds
|
Transitions time can be from 0 to 60 seconds with a default of 10.
|
Command Default
10 seconds.
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
Because the available circuit (AC) attribute of a destination is very dynamic, reporting of this attribute should be handled carefully. AC should be reported as frequently as possible so that the location server has better information about the resources. However, the location server should not be overwhelmed with too many updates.
A smoothing algorithm is applied to the quantity of AC being reported. This algorithm eliminates reporting of noise. The degree of smoothing can be configured with the voice call capacity stw command. This command sets the smoothing transition time for weight, which is the time it takes for current smoothed value of AC to come half way between the current smoothed value and the current instantaneous value of AC. Lower stw values speed the smoothed value of AC as it approaches the instantaneous value of AC. When stw is set to 0, the smoothed value is always equal to the instantaneous value of AC.
Examples
The following example shows the smoothing time for weight for the carrier address family set to 25 seconds:
Router(config)# voice call carrier capacity stw 25
Related Commands
Command
|
Description
|
capacity update interval (dial peer)
|
Changes the capacity update for prefixes associated with a dial peer.
|
capacity update interval (trunk group)
|
Change the capacity update for carriers or trunk groups.
|
voice call capacity mir
|
Set the value for MIR.
|
voice call capacity timer interval
To set the periodic interval for reporting capacity from carrier, trunk group, or prefix databases, use the voice call capacity timer interval command in global configuration mode. To turn off the interval, use the no form of this command.
voice call {carrier | trunk-group | prefix} capacity timer interval seconds
no voice call {carrier | trunk-group | prefix} capacity timer interval seconds
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
seconds
|
Value from 10 to 3600 seconds.
|
Command Default
25 seconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
For the reporting interval, a periodic timer called the capacity update timer handles updates of available circuit (AC) information and can be configured using the voice call capacity timer interval command. For example, if AC has changed since the last reporting, the AC is again reported when the capacity update timer expires.
Examples
The following example sets the timer interval for the prefixes set at 15 seconds:
Router(config)# voice call prefix capacity timer interval 15
Related Commands
Command
|
Description
|
voice call capacity mir
|
Sets the values for the MIR and STW.
|
voice call capacity reporting
|
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
|
voice call trigger hwm
|
Sets the value for percentage change, low water mark and high water mark in the available capacity in the trunk group or prefix databases.
|
voice call convert-discpi-to-prog
To convert a disconnect message with a progress indicator (PI) to a progress message, use the voice call convert-discpi-to-prog command in global configuration mode. To return to the default condition, use the no form of this command.
voice call convert-discpi-to-prog [tunnel-IEs | always [tunnel-IEs]]
no voice call convert-discpi-to-prog
Syntax Description
tunnel-IEs
|
(Optional) Information elements (IEs) are carried in the progress message.
|
always
|
(Optional) Converts disconnect message with a PI to a progress message in both preconnected and connected states.
|
Command Default
A disconnect message with a PI is not converted to a progress message.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(1)
|
This command was introduced.
|
12.3(6)
|
The tunnel-1Es keyword was added.
|
12.3(4)XQ
|
The always keyword with the tunnel-IEs keyword were added.
|
12.3(8)T
|
The always keyword with the tunnel-IEs keyword were added.
|
12.3(9)
|
The always keyword with the tunnel-1Es keyword were added.
|
Usage Guidelines
The voice call convert-discpi-to-prog command turns an ISDN disconnect message into a progress message. If you use the tunnel-IEs keyword, the information elements are not dropped when the disconnect message is converted to a progress message.
Examples
The following example changes a disconnect with PI to a progress message containing information elements (IEs):
voice call convert-discpi-to-prog tunnel-IEs
The following example changes a disconnect with PI to a progress message in the preconnected and connected states:
voice call convert-discpi-to-prog always
Related Commands
Command
|
Description
|
disc_pi_off
|
Enables an H.323 gateway to disconnect a call when it receives a disconnect message with a PI.
|
voice call csr data-points
To set the number of call success rate (CSR) data points, use the voice call csr data-points command in global configuration mode. To disable the setting of the CSR data points, use the no form of this command.
voice call {carrier | trunk-group | prefix} csr data-points value
no voice call {carrier | trunk-group | prefix} csr data-points value
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
value
|
Value from 10 to 50 data points. Default is 30 data points.
|
Command Default
30 data points
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Examples
The following example sets the CSR data points for trunk groups at 10:
Router(config)# voice call trunk-group csr data-points 10
Related Commands
Command
|
Description
|
voice call csr recording interval
|
Sets the recording interval for CSR.
|
voice call csr reporting interval
|
Sets the reporting interval for CSR.
|
voice call csr recording interval
To set the recording interval for call success rates (CSR), use the voice call csr recording interval command in global configuration mode. To disable the CSR recording interval, use the no form of this command.
voice call {carrier | trunk-group | prefix} csr recording interval minutes
no voice call {carrier | trunk-group | prefix} csr recording interval minutes
Syntax Description
carrier
|
Carrier code address family.
|
trunk-group
|
Trunk group address family.
|
prefix
|
E.164 prefix.
|
minutes
|
Value from 10 to 1000 minutes with a default of 60.
|
Command Default
60 minutes
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Examples
The following example sets the CSR recording interval for prefixes at 30 minutes:
Router(config)# voice call carrier csr recording interval 30
Related Commands
Command
|
Description
|
voice call csr data-points
|
Sets the number of call success rate (CSR) data points.
|
voice call csr reporting interval
|
Sets the reporting interval for CSR.
|
voice call csr reporting interval
To set the reporting interval for call success rate (CSR), use the voice call csr reporting interval command in global configuration mode. To disable the CSR recording interval, use the no form of this command.
voice call {carrier | trunk-group | prefix} csr reporting interval seconds
no voice call {carrier | trunk-group | prefix} csr reporting interval seconds
Syntax Description
carrier
|
Carrier code address family.
|
trunk-group
|
Trunk group address family.
|
prefix
|
E.164 prefix.
|
seconds
|
Value from 10 to 10000 seconds with a default of 25.
|
Command Default
25 seconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Examples
The following example sets the CSR reporting interval for trunk groups at 40 seconds:
Router(config)# voice call carrier csr reporting interval 40
Related Commands
Command
|
Description
|
voice call csr data-points
|
Sets the number of CSR data points.
|
voice call csr recording interval
|
Sets the recording interval for CSR.
|
voice call debug
To debug a voice call, use the voice call debug command in global configuration mode. To display a full globally unique identifier (GUID) or header as explained in the Usage Guidelines section, use the no form of this command.
voice call debug full-guid | short-header
no voice call debug full-guid | short-header
Syntax Description
full-guid
|
Displays the GUID in a 16-byte header.
Note When the no version of this command is input with the full-guid keyword, the short 6-byte version displays. This is the default.
|
short-header
|
Displays the CallEntry ID in the header without displaying the GUID or module-specific parameters.
|
Command Default
The short 6-byte header displays.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
The new debug header was added to the following platforms: Cisco 2600 series, Cisco 3620, Cisco 3640, Cisco 3660 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, Cisco AS5850, and Cisco MC3810.
|
12.2(15)T
|
The header-only argument was removed and the short-header argument was added.
|
Usage Guidelines
Despite its nontraditional syntax (trailing rather than preceding "debug"), this is a normal debug command.
You can control the contents of the standardized header. Display options for the header are as follows:
•Short 6-byte GUID
•Full 16-byte GUID
•Short header which contains only the CallEntry ID
The format of the GUID headers is as follows:
//CallEntryID/GUID/Module-Dependent-List/Function-name:.
The format of the short header is as follows:
//CallEntryID/Function-name:.
When the voice call debug short-header command is entered, the header displays with no GUID or module-specific parameters. When the no voice call debug short-header command is entered, the header, the 6-byte GUID, and module-dependent parameter output displays. The default option is displaying the 6-byte GUID trace.
Note Using the no form of this command does not turn off debugging.
Examples
The following is sample output when the full-guid keyword is specified:
Router# voice call debug full-guid
00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_insert_cdb:
00:05:12: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/CCAPI/cc_incr_if_call_volume:
00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_open_voice_and
_set_params:
00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_modem_proto_fr
om_cdb:
00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/set_playout_cdb:
00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_dsp_echo_cance
ller_control:
Note The "//-1/" output indicates that CallEntryID for the CCAPI module is not available.
Table 240 describes significant fields shown in the display.
Table 240 voice call debug full-guid Field Descriptions
Field
|
Description
|
VTSP:(0:D):0:0:4385
|
VTSP module, port name, channel number, DSP slot, and DSP channel number.
|
vtsp_insert_cdb
|
Function name.
|
CCAPI
|
CCAPI module.
|
The following is sample output when the short-header keyword is specified:
Router(config)# voice call debug short-header
00:05:12: //1/vtsp_insert_cdb:
00:05:12: //-1/cc_incr_if_call_volume:
00:05:12: //1/vtsp_open_voice_and_set_params:
00:05:12: //1/vtsp_modem_proto_from_cdb:
00:05:12: //1/set_playout_cdb:
00:05:12: //1/vtsp_dsp_echo_canceller_control:
Note The "//-1/" output indicates that CallEntryID for CCAPI is not available.
Related Commands
Command
|
Description
|
debug rtsp api
|
Displays debug output for the RTSP client API.
|
debug rtsp client session
|
Displays debug output for the RTSP client data.
|
debug rtsp error
|
Displays error message for RTSP data.
|
debug rtsp pmh
|
Displays debug messages for the PMH.
|
debug rtsp socket
|
Displays debug output for the RTSP client socket data.
|
debug voip ccapi error
|
Traces error logs in the CCAPI.
|
debug voip ccapi inout
|
Traces the execution path through the CCAPI.
|
debug voip ivr all
|
Displays all IVR messages.
|
debug voip ivr applib
|
Displays IVR API libraries being processed.
|
debug voip ivr callsetup
|
Displays IVR call setup being processed.
|
debug voip ivr digitcollect
|
Displays IVR digits collected during the call.
|
debug voip ivr dynamic
|
Displays IVR dynamic prompt play debug.
|
debug voip ivr error
|
Displays IVR errors.
|
debug voip ivr script
|
Displays IVR script debug.
|
debug voip ivr settlement
|
Displays IVR settlement activities.
|
debug voip ivr states
|
Displays IVR states.
|
debug voip ivr tclcommands
|
Displays the TCL commands used in the script.
|
debug voip rawmsg
|
Displays the raw VoIP message.
|
debug vtsp all
|
Enables debug vtsp session, debug vtsp error, and debug vtsp dsp.
|
debug vtsp dsp
|
Displays messages from the DSP.
|
debug vtsp error
|
Displays processing errors in the VTSP.
|
debug vtsp event
|
Displays the state of the gateway and the call events.
|
debug vtsp port
|
Limits VTSP debug output to a specific voice port.
|
debug vtsp rtp
|
Displays the voice telephony RTP packet debugging.
|
debug vtsp send-nse
|
Triggers the VTSP software module to send a triple redundant NSE.
|
debug vtsp session
|
Traces how the router interacts with the DSP.
|
debug vtsp stats
|
Debugs periodic statistical information sent and received from the DSP
|
debug vtsp vofr subframe
|
Displays the first 10 bytes of selected VoFR subframes for the interface.
|
debug vtsp tone
|
Displays the types of tones generated by the VoIP gateway.
|
voice call disc-pi-off
To enable the gateway to treat a disconnect message with progress indicator (PI) like a standard disconnect without a PI, use the voice call disc-pi-off command in global configuration mode. To reset to the default, use the no form of this command.
voice call disc-pi-off
no voice call disc-pi-off
Syntax Description
This command has no keywords or arguments.
Command Default
Gateway disconnects incoming call leg when it receives a disconnect message with PI.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(5)
|
This command was introduced.
|
12.3(7)T
|
This command was integrated into Cisco IOS Release 12.3(7)T.
|
Usage Guidelines
Use this command if the gateway is connected to a switch that sends a release immediately after it receives a Disconnect with PI. To properly handle the call, the switch should open a backward voice path and keep the call active. Otherwise the rotary dial peer feature does not work because the incoming call leg is disconnected. Using this command enables the gateway to handle a disconnect with PI like a regular disconnect message so that you can use the rotary dial peer feature.
Examples
The following example enables the gateway to properly handle a disconnect with PI:
Related Commands
Command
|
Description
|
disc_pi_off
|
Enables an H.323 gateway to disconnect a call when it receives a disconnect message with a PI.
|
voice call convert-discpi-to-prog
|
Converts a disconnect message with a PI to a progress message.
|
voice call send-alert
To enable the terminating gateway to send an alert message instead of a progress message after it receives a call setup message, use the voice call send-alert command in global configuration mode. To reset to the default, use the no form of this command.
voice call send-alert
no voice call send-alert
Syntax Description
This command has no arguments or keywords.
Command Default
The terminating gateway sends a progress message after it receives a call Setup message.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)XI4
|
This command was introduced.
|
12.1(5)T
|
This command was not supported in this release.
|
12.1(5.3)T
|
This command was integrated into Cisco IOS Release 12.1(5.3)T.
|
12.2(1)
|
This command was integrated into Cisco IOS Release 12.2.
|
Usage Guidelines
In Cisco IOS Release 12.1(3)XI and later, the terminating gateway sends a Progress message with a progress indicator (PI) after it receives a Setup message. Previously, the gateway responded with an Alert message after receiving a call. In some cases, if the terminating switch does not forward the progress message to the originating gateway, the originating gateway does not cut-through the voice path until a Connect is received and the caller does not hear a ringback tone. In these cases, you can use the voice call send-alert command to make the gateway backward compatible with releases earlier than Cisco IOS Release 12.1(3)XI. If you configure the voice call send-alert command, the terminating gateway sends an Alert message after it receives a Setup message from the originating gateway.
To complete calls from a PRI to an FXS interface, configure the voice call send-alert command on the FXS device.
Examples
The following example configures the gateway to send an Alert message:
Related Commands
Command
|
Description
|
progress_ind
|
Sets a specific PI in call Setup, Progress, or Connect messages from an H.323 VoIP gateway.
|
voice call trigger hwm
To set the value for high water mark in the available capacity in the trunk group or prefix databases, use the voice call trigger hwm command in global configuration mode. To disable the trigger point, use the no form of this command.
voice call {carrier | trunk-group | prefix} trigger hwm percent
no voice call {carrier | trunk-group | prefix} trigger hwm percent
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
percent
|
Value can be 50 to 100 percent with a default of 80. If set to 100, this trigger will be turned off.
|
Command Default
80 percent
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
Available circuits are reported when the value of AC goes above a threshold, called the high water mark. This can be configured with the voice call trigger hwm command. When the hwm option is selected and the value is set to 100, no update is sent due to high water mark.
Examples
The following example sets the trigger for available capacity on trunk groups to send at a high water mark of 75%:
Router(config)# voice call trunk-group trigger hwm 75
Related Commands
Command
|
Description
|
voice call capacity mir
|
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
|
voice call capacity reporting
|
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
|
voice call capacity timer interval
|
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases
|
voice call trigger lwm
|
Sets the value for low water mark in the available capacity for carrier, trunk group, or prefix databases
|
voice call trigger percent-change
|
Sets the value for percentage change in the available capacity for carrier, trunk group, or prefix databases
|
voice call trigger lwm
To set the value for low water mark in the available capacity in the trunk group or prefix databases, use the voice call trigger lwm command in global configuration mode. To disable the trigger point, use the no form of this command.
voice call {carrier | trunk-group | prefix} trigger lwm percent
no voice call {carrier | trunk-group | prefix} trigger lwm percent
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
percent
|
Value can be 0 to 30 percent with a default of 10. If set to 0, this trigger will be turned off.
|
Command Default
10 percent
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
Available circuits are reported when the value of AC falls below a threshold, called the low water mark. When the lwm option is selected and the value is set to 0, no update is sent due to low water mark.
Examples
The following example sets the trigger for available capacity for E.164 prefixes to send at a low water mark of 25%:
Router(config)# voice call prefix trigger lwm 25
Related Commands
Command
|
Description
|
voice call capacity mir
|
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
|
voice call capacity reporting
|
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
|
voice call capacity timer interval
|
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases.
|
voice call trigger hwm
|
Sets the value for high water mark in the available capacity for carrier, trunk group, or prefix databases
|
voice call trigger percent-change
|
Sets the value for percentage change in the available capacity for carrier, trunk group, or prefix databases
|
voice call trigger percent-change
To set the value for percentage change, low water mark and high water mark in the available capacity in the trunk group or prefix databases, use the voice call trigger command in global configuration mode. To disable the trigger point, use the no form of this command.
voice call {carrier | trunk-group | prefix} trigger percent-change percent
no voice call {carrier | trunk-group | prefix} trigger percent-change percent
Syntax Description
carrier
|
Carrier code address family
|
trunk-group
|
Trunk group address family
|
prefix
|
E.164 prefix
|
percent
|
If percent-change is selected, value can be 0 to 100 percent with a default of 30. If set to 0, this trigger will be turned off.
If lwm is selected, value can be 0 to 30 percent with a default of 10. If set to 0, this trigger will be turned off.
If hwm is select, value can be 50 to 100 percent with a default of 80. If set to 100, this trigger will be turned off.
|
Command Default
30 percent
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.3(1)
|
This command was introduced.
|
Usage Guidelines
Available circuits are reported when the absolute percent change is above a threshold. When the percent-change option is selected and the value is set to 0, no update for percent change is sent
Examples
The following example sets the trigger for available capacity on the carrier codes to send at a percentage change of 15%:
Router(config)# voice call carrier trigger percent-change 15
Related Commands
Command
|
Description
|
voice call capacity mir
|
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
|
voice call capacity reporting
|
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
|
voice call capacity timer interval
|
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases
|
voice call trigger hwm
|
Sets the value for high water mark in the available capacity for carrier, trunk group, or prefix databases
|
voice call trigger lwm
|
Sets the value for low water mark in the available capacity for carrier, trunk group, or prefix databases
|
voicecap configure
To apply a voicecap on NextPort platforms, use the voicecap configure command in voice-port configuration mode. To remove a voicecap, use the no form of this command.
voicecap configure {name}
no voicecap configure {name}
Syntax Description
name
|
Designates which voicecaps to use on this voice port.
|
Command Default
No default values or behavior
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
Usage Guidelines
The character value for the name argument must be identical to the value entered when you created the voicecap using the voicecap entry command.
Examples
The following example configures a voicecap with the name qualityERL:
Router# configure terminal
Router(config)# voicecap entry qualityERL v270=120
Router(config)# voice-port 3/0:D
Router(config-voiceport)# voicecap configure qualityERL
Related Commands
Command
|
Description
|
voicecap entry
|
Creates a voicecap on NextPort platforms.
|
voicecap entry
To create a voicecap, use the voicecap entry command in global configuration mode. To disable a voicecap, use the no form of this command.
voicecap entry [name string]
no voicecap entry [name string]
Syntax Description
name string
|
(Optional) A word and a string of characters that uniquely identify a voicecap.
•The name argument specifies a unique identifier for a voicecap.
•The string argument specifies one or more voicecap register entries, similar to a modemcap. Each entry is of the form vindex=value, where index refers to a specific V register, and value designates the value for that V register.
|
Command Default
No voice caps can be applied to configure firmware.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.3(4)T
|
This command was introduced.
|
12.3(11)T
|
This command was integrated into Cisco IOS Release 12.3(11)T.
|
12.4(4)XC
|
This command was modified to include GSMAMR-NB codec capability.
|
12.4(9)T
|
This command was integrated into Cisco IOS Release 12.4(9)T.
|
Usage Guidelines
This command configures firmware through voicecap strings. This command allows you to assign values to specific registers. Voicecaps are applied to specific voice ports at system startup.
The voicecap values can be entered in a DSP-recognizable format called raw format. They can also be entered in standard format, which allows you to use commonly accessible values, such as decibels.
Starting with Cisco IOS Release 12.4(4)XC, this command can be used to configure GSMAMR-NB codecs on Cisco AS5350XM and Cisco AS5400XM platforms. The register values for GSMAMR-NB are shown in Table 241.
Table 241 GSMAMR-NB Register Values
V-Reg #
|
Default
|
Description
|
Register Values and Additional Notes
|
0
|
0
|
Sets how Adaptive Multi-Rate (AMR) responds to an incoming codec mode request (CMR) that is not a member of the mode set.
|
0 = Drop the packet with the bad CMR. 1 = Ignore the CMR (do not change rates) but process the rest of the packet data normally. 2 = Change the rate to the highest rate in the mode set lower than the rate requested by the CMR.
|
1
|
0
|
Sets how AMR handles packets with a frame type (AMR rate) that is not a member of the mode set.
|
0 = Drop the packet with the bad frame-type. 1 = Attempt to decode the packet.
|
Examples
The following example creates a voicecap string for a GSMAMR-NB codec named gsmamrnb-ctrl with V register 0 set to 1:
Router# configure terminal
Router(config)# voicecap entry gsmamrnb-ctrl v0=1
Related Commands
Command
|
Description
|
voicecap configure
|
Applies a voicecap to the specified voice ports.
|
voice-card
To enter voice-card configuration mode and configure a voice card, use the voice-card command in global configuration mode. There is no no form of this command.
voice-card slot
Syntax Description
slot
|
Slot number for the card to be configured. The following platform-specific numbering schemes apply:
•Cisco 2600 series and Cisco 2600XM:
–0 is the Advanced Integration Module (AIM) slot in the router chassis.
–1 is the network module slot in the router chassis.
•Cisco 3600 series:
–A value from 1 to 6 identifies a network module slot in the router chassis.
•Cisco 3660:
–7 is AIM slot 0 in the router chassis.
–8 is AIM slot 1.
|
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)XK
|
The command was introduced on the Cisco 2600 series and Cisco 3600 series.
|
12.0(7)T
|
This command was integrated into Cisco IOS Release 12.0(7)T.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.2(2)XB
|
Values for the slot argument were updated to include AIMs.
|
12.2(8)T
|
This command was integrated into Cisco IOS Release 12.2(8)T.
|
12.2(13)T
|
This command was supported in Cisco IOS Release 12.2(13)T and implemented on the Cisco 1700 series, Cisco 2600XM, Cisco 3700 series, Cisco 7200 series, Cisco 7500 series, Cisco ICS7750, Cisco MC3810, and Cisco VG200.
|
12.2(15)T
|
This command was integrated into Cisco IOS Release 12.2(15)T.
|
Usage Guidelines
Voice-card configuration mode is used for commands that configure the use of digital signal processing (DSP) resources, such as codec complexity and DSPs. DSP resources can be found in digital T1/E1 packet voice trunk network modules on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series.
Codec complexity is configured in voice-card configuration mode and has the following platform-specific usage guidelines:
•On Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745, the slot argument corresponds to the physical chassis slot of the network module that has DSP resources to be configured.
DSP resource sharing is also configured in voice-card configuration mode. On the Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745 under specific circumstances, configuration of the dspfarm command enters DSP resources on a network module or AIM into a DSP resource pool. Those DSP resources are then available to process voice traffic on a different network module or voice/WAN interface card (VWIC). See the dspfarm (voice-card) command reference for more information about DSP resource sharing.
Note When running high-complexity images, the system can only process up to 16 voice channels. Those 16 time slots need to be within a contiguous range (timeslot maximum (TSmax) minus timeslot minimum (TSmin) is less than or equal to 16, where TSmax and TSmin are the maximum DS0 and minimum DS0 configured for voice).
This command does not have a no form.
Examples
The following example enters voice-card configuration mode to configure resources on the network module in slot 1:
The following example shows how to enter voice-card configuration mode and load high-complexity DSP firmware on voice-card 0. The dspfarm command enters the DSP resources on the AIM specified in the voice-card command into the DSP resource pool.
Related Commands
Command
|
Description
|
codec complexity
|
Matches the DSP complexity packaging to the codecs to be supported.
|
dspfarm (voice-card)
|
Adds the specified voice card to those participating in a DSP resource pool.
|
voice class aaa
To enable dial-peer-based VoIP AAA configurations, use the voice class aaa command in global configuration mode. To disable dial-peer-based VoIP AAA configurations, use the no form of this command.
voice class aaa tag
no voice class aaa tag
Syntax Description
tag
|
A number used to identify voice class AAA. The range is from 1 to 10000. There is no default value.
|
Command Default
No default behaviors or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
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Usage Guidelines
The voice class aaa configuration command sets up a voice service class that allows you to perform dial-peer-based AAA configurations.
The command activates voice class AAA configuration mode. Commands that are configured in voice class AAA configuration mode are listed in the "Related Commands" section.
Examples
The following example shows AAA configurations in voice class AAA configuration mode. The number assigned to the tag is 1.
accounting template temp-dp
The following example shows accounting configurations in voice class AAA configuration mode:
accounting method dp-out out-bound
accounting template temp-dp out-bound
Related Commands
Command
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Description
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authentication method
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Specifies an authentication method for calls coming into the defined dial peer.
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authorization method
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Specifies an authorization method for calls coming into the defined dial peer.
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method
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Specifies an accounting method for calls coming into the defined dial peer.
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accounting suppress
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Disables accounting that is automatically generated by the service provider module for a specific dial peer.
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voice-class aaa
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Applies properties defined in the voice class to a specific dial peer.
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voice-class aaa (dial peer)
To apply properties defined in the voice class to a dial peer, use the voice-class aaa command in dial peer configuration mode. This command does not have a no form.
voice-class aaa tag
Syntax Description
tag
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A number to identify the voice class. Range is from 1 to 10000. There is no default.
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Command Default
No default behaviors or values
Command Modes
Dial peer configuration
Command History
Release
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Modification
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12.2(11)T
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This command was introduced on the Cisco 3660, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
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Usage Guidelines
Properties that are configured in voice class AAA configuration mode can be applied to a dial peer by using this command.
Examples
The following example shows redirecting AAA requests using Digital Number Identification Service (DNIS). You define a voice class to specify the AAA methods and then use this command.
incoming called-number 50..
session target ipv4:1.5.31.201
Related Commands
Command
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Description
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voice class aaa
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Enables dial-peer-based VoIP AAA configurations.
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voice class busyout
To create a voice class for local voice busyout functions, use the voice class busyout command in global configuration mode. To delete the voice class, use the no form of this command.
voice class busyout tag
no voice class busyout tag
Syntax Description
tag
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Unique identification number assigned to one voice class. Range is 1 to 10000.
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Command Default
No voice class is configured for busyout functions.
Command Modes
Global configuration
Command History
Release
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Modification
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12.1(3)T
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This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
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Usage Guidelines
You can apply a busyout voice class to multiple voice ports. You can assign only one busyout voice class to a voice port. If a second busyout voice class is assigned to a voice port, the second voice class replaces the one previously assigned.
If you assign a busyout voice class to a voice port, you may not assign separate busyout commands directly to the voice port, such as busyout monitor serial, busyout monitor ethernet, or busyout monitor probe.
Examples
The following example configures busyout voice class 20, in which the connections to two remote interfaces are monitored by a response time reporter (RTR) probe with a G.711ulaw profile, and voice ports are busied out whenever both links have a packet loss exceeding 10 percent and a packet delay time exceeding 2 seconds:
busyout monitor probe 171.165.202.128 g711u loss 10 delay 2000
busyout monitor probe 171.165.202.129 g711u loss 10 delay 2000
The following example configures busyout voice class 30, in which voice ports are busied out when serial ports 0/0, 1/0, 2/0, and 3/0 go out of service.
busyout monitor serial 0/0
busyout monitor serial 1/0
busyout monitor serial 2/0
busyout monitor serial 3/0
Related Commands
Command
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Description
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busyout monitor ethernet
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Configures a voice port to monitor a local Ethernet interface for events that would trigger a voice-port busyout.
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busyout monitor probe
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Configures a voice port to enter the busyout state if an RTR probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold.
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busyout monitor serial
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Configures a voice port to monitor a serial interface for events that would trigger a voice-port busyout.
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show voice busyout
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Displays information about the voice busyout state.
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voice-class called-number (dial peer)
To assign a previously defined voice class called number to an inbound or outbound POTS dial peer, use the voice-class called-number command in dial peer configuration mode. To remove a voice class called number from the dial peer, use the no form of this command.
voice-class called-number [inbound | outbound] tag
no voice-class called-number
Syntax Description
inbound
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Assigns an inbound voice class called number to the dial peer.
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outbound
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Assigns an outbound voice class called number to the dial peer.
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tag
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Digits that identify a specific voice class called number.
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Command Default
No voice class called number is configured on the dial peer.
Command Modes
Dial peer configuration
Command History
Release
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Modification
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12.4(11)T
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This command was introduced.
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Usage Guidelines
Use this command to assign a previously defined voice class called number to a dial peer for a static H.320 secondary call dial plan. Use the inbound keyword for inbound POTS dial peers, and the outbound keyword for outbound POTS dial peers.
Note The voice class called number command in global configuration mode is entered without hyphens. The voice-class called-number command in dial peer configuration mode is entered with hyphens.
Examples
The following example shows configuration for an outbound voice class called number outbound on POTS dial peer 22:
voice-class called-number inbound 300
Related Commands
Command
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Description
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voice class called number
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Defines a voice class called number or range of numbers for H.320 calls.
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voice-class called-number-pool
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Defines a pool of dynamic voice class called numbers for a voice port.
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voice class called number
To define a voice class called number or range of numbers, use the voice class called number command in global configuration mode. To remove a voice class called number, use the no form of this command.
voice class called number {inbound | outbound | pool} tag
no voice class called number
Syntax Description
inbound
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Inbound voice class called number.
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outbound
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Outbound voice class called number.
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pool
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Voice class called number pool.
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tag
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Digits that identify a specific inbound or outbound voice class called number or voice class called number pool.
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Command Default
No voice class called number is configured.
Command Modes
Global configuration
Command History
Release
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Modification
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12.4(11)T
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This command was introduced.
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Usage Guidelines
Use this command to define one or more static voice class called numbers for inbound and outbound POTS dial peers or a dynamic voice class called number pool. The indexes for a voice class called number are defined with the index (voice class) command.
Note Enter the voice class called number command in global configuration mode without hyphens. Enter the voice-class called-number command in dial peer configuration mode with hyphens.
Examples
The following example shows configuration for an outbound voice class called number:
voice class called number outbound 30
The following example shows configuration for a voice class called number pool:
voice class called number pool 1
index 1 5550100 - 5550199
Related Commands
Command
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Description
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show voice class called-number
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Displays a specific voice class called number.
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voice-class called-number (dial peer)
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Assigns a previously defined voice class called number to an inbound or outbound POTS dial peer.
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voice-class called-number-pool
To assign a previously defined voice class called number pool to a voice port, use the voice-class called-number-pool command in voice class configuration mode. To remove a voice class called number pool from the voice port, use the no form of this command.
voice-class called-number-pool tag
no voice-class called-number-pool
Syntax Description
tag
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Digits that identify a specific voice class called number pool.
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Command Default
No voice class called number pool is assigned to the voice port.
Command Modes
Voice class configuration
Command History
Release
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Modification
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12.4(11)T
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This command was introduced.
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Usage Guidelines
Use this command to assign a voice class called number pool to a voice port for a dynamic H.320 secondary call dial plan.
Examples
The following example shows configuration for voice class called number pool 100 on voice port 1/0/0:
voice-class called-number-pool 100
Related Commands
Command
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Description
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voice class called number
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Defines a voice class called number or range of numbers for H.320 calls.
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voice-class called-number (dial peer)
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Defines a called number or range of called numbers for a POTS dial peer.
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voice class codec
To enter voice-class configuration mode and assign an identification tag number for a codec voice class, use the voice class codec command in global configuration mode. To delete a codec voice class, use the no form of this command.
voice class codec tag
no voice class codec tag
Syntax Description
tag
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Unique number that you assign to the voice class. Range is from 1 to 10000. There is no default.
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Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Release
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Modification
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12.0(2)XH
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This command was introduced on the Cisco AS5300.
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12.0(7)T
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This command was implemented on the Cisco 2600 series and Cisco 3600 series.
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12.0(7)XK
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This command was implemented on the Cisco MC3810.
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12.1(2)T
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This command was integrated into Cisco IOS Release 12.1(2)T.
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Usage Guidelines
This command only creates the voice class for codec selection preference and assigns an identification tag. Use the codec preference command to specify the parameters of the voice class, and use the voice-class codec dial-peer command to apply the voice class to a VoIP dial peer.
Note The voice class codec command in global configuration mode is entered without a hyphen. The voice-class codec command in dial peer configuration mode is entered with a hyphen.
Examples
The following example shows how to enter voice-class configuration mode and assign a voice class tag number starting from global configuration mode:
After you enter voice-class configuration mode for codecs, use the codec preference command to specify the parameters of the voice class.
The following example creates preference list 99, which can be applied to any dial peer:
codec preference 1 g711alaw
codec preference 2 g711ulaw bytes 80
codec preference 3 g723ar53
codec preference 4 g723ar63 bytes 144
codec preference 5 g723r53
codec preference 6 g723r63 bytes 120
codec preference 7 g726r16
codec preference 8 g726r24
codec preference 9 g726r32 bytes 80
codec preference 11 g729br8
codec preference 12 g729r8 bytes 50
Related Commands
Command
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Description
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codec preference
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Specifies a list of preferred codecs to use on a dial peer.
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test voice port detector
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Defines the order of preference in which network dial peers select codecs.
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voice-class codec (dial peer)
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Assigns a previously configured codec selection preference list to a dial peer.
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voice-class codec (dial peer)
To assign a previously configured codec selection preference list (codec voice class) to a Voice over IP (VoIP) dial peer, enter the voice-class codec command in dial peer configuration mode. To remove the codec preference assignment from the dial peer, use the no form of this command.
voice-class codec tag
no voice-class codec tag
Syntax Description
tag
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Unique number assigned to the voice class. Range is from 1 to 10000. The tag number maps to the tag number created using the voice class codec global configuration command.
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Command Default
Dial peers have no codec voice class assigned.
Command Modes
Dial peer configuration
Command History
Release
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Modification
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12.0(2)XH
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This command was introduced on Cisco AS5300.
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12.0(7)T
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This command was supported on Cisco 2600 series and Cisco 3600 series.
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12.0(7)XK
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This command was supported on Cisco MC3810.
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12.1(2)T
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This command was integrated into Cisco IOS Release 12.1(2)T.
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Usage Guidelines
You can assign one voice class to each VoIP dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
Note The voice-class codec command in dial peer configuration mode is entered with a hyphen. The voice class codec command in global configuration mode is entered without a hyphen.
Examples
The following example shows how to assign a previously configured codec voice class to a dial peer:
Related Commands
Command
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Description
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show dial-peer voice
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Displays the configuration for all dial peers configured on the router.
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test voice port detector
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Defines the order of preference in which network dial peers select codecs.
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voice class codec
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Enters voice-class configuration mode and assigns an identification tag number for a codec voice class.
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voice class custom-cptone
To create a voice class for defining custom call-progress tones to be detected, use the voice class custom-cptone command in global configuration mode. To delete the voice class, use the no form of this command.
voice class custom-cptone cptone-name
no voice class custom-cptone cptone-name
Syntax Description
cptone-name
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Descriptive identifier for this class of custom call-progress tones that associates this set of custom call-progress tones with voice ports.
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Command Default
No voice class of custom call-progress tones is created.
Command Modes
Global configuration
Command History
Release
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Modification
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12.1(5)XM
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This command was introduced on the Cisco 2600, Cisco 3600, and Cisco MC3810 platforms.
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12.2(2)T
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This command was implemented on Cisco 1750 access routers and integrated into Cisco IOS Release 12.2(2)T.
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Usage Guidelines
After you create a voice class, you need to define custom call-progress tones for this voice class using the dualtone command.
Examples
The following example creates a voice class named country-x.
voice class custom-cptone country-x
The following example deletes the voice class named country-x.
no voice class custom-cptone country-x
Related Commands
Command
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Description
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dualtone
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Defines the tone and cadence for a custom call-progress tone.
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supervisory custom-cptone
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Associates a class of custom call-progress tones with a voice port.
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voice class dualtone-detect-params
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Modifies the boundaries and limits for call-progress tones.
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voice class dualtone
To create a voice class for Foreign Exchange Office (FXO) supervisory disconnect tone detection parameters, use the voice class dualtone command in global configuration mode. To delete the voice class, use the no form of this command.
voice class dualtone tag
no voice class dualtone tag
Syntax Description
tag
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Unique identification number assigned to one voice class. Range is from 1 to 10000.
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Command Default
No voice class is configured for tone detection parameters.
Command Modes
Global configuration
Command History
Release
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Modification
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12.1(3)T
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This command was introduced on the Cisco 2600 series, Cisco 3600, and the Cisco MC3810.
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Usage Guidelines
Use this command first to create the voice class. Then use the supervisory disconnect dualtone voice-class command to assign the voice class to a voice port.
A voice class can define any number of tones to be detected. You need to define a matching tone for each supervisory disconnect tone expected from a PBX or from the public switched telephone network (PSTN).
Examples
The following example configures voice class dualtone 70, which defines one tone with two frequency components, and does not configure a cadence list:
The following example configures voice class dualtone 100, which defines one tone with two frequency components, and configures a cadence list:
cadence-list 1 100 100 300 300
The following example configures voice class dualtone 90, which defines three tones, each with two frequency components, and configures two cadence lists:
cadence-list 1 100 100 300 300 100 200
cadence-list 2 100 200 100 400
Related Commands
Command
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Description
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supervisory disconnect dualtone voice-class
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Assigns a previously configured voice class for FXO supervisory disconnect tone to a voice port.
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voice class dualtone-detect-params
To create a voice class for defining a set of tolerance limits for the frequency, power, and cadence parameters of the tones to be detected, use the voice class dualtone-detect-params command in global configuration mode. To delete the voice class, use the no form of this command.
voice class dualtone-detect-params tag
no voice class dualtone-detect-params tag
Syntax Description
tag
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Unique tag identification number assigned to a voice class. Range is from 1 to 10000.
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Command Default
No voice class is configured for defining answer-supervision tolerance limits.
Command Modes
Global configuration
Command History
Release
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Modification
|
12.1(5)XM
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
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12.2(2)T
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This command was implemented on Cisco 1750 routers and integrated into Cisco IOS Release 12.2(2)T.
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Usage Guidelines
Use this command to create a voice class in which you can define maximum and minimum call-progress tone tolerance parameters that you can apply to any voice port. These parameters further define the call-progress tones defined by the voice class custom-cptone command. Use the supervisory dualtone-detect-params command to apply these tolerance parameters to a voice port.
Examples
The following example creates voice class 70, in which you can specify modified boundaries and limits for call-progress tone detection.
voice class dualtone-detect-params 70
Related Commands
Command
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Description
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supervisory dualtone-detect-params
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Assigns the boundary and detection tolerance parameters defined by the voice class dualtone-detect-params command to a voice port.
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voice class custom-cptone
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Creates a voice class for defining custom call-progress tones.
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