Table Of Contents
Cisco IOS Voice Commands:
V through W
vad (dial peer)
vad (voice-port)
vbd-playout-delay maximum
vbd-playout-delay minimum
vbd-playout-delay mode
vbd-playout-delay nominal
vbr-rt
vcci
vm-device-id (ephone)
vm-integration
vofr
voice
voice call capacity mir
voice call capacity stw
voice call capacity reporting
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 send-alert
voice call trigger hwm
voice call trigger lwm
voice call trigger percent-change
voice class aaa
voice-class aaa (dial peer)
voice class busyout
voice class codec
voice-class codec (dial peer)
voice class custom-cptone
voice class dualtone
voice class dualtone-detect-params
voice class h323
voice-class h323 (dial peer)
voice class permanent
voice-class permanent (dial-peer)
voice-class permanent (voice-port)
voice confirmation-tone
voice dnis-map
voice dnis-map load
voice echo-canceller extended
voice enum-match-table
voice hpi capture
voice hunt
voice local-bypass
voice rtp send-recv
voice service
voice source-group
voice translation-profile
voice translation-rule
voice vad-time
voice-card
voice-class sip rel1xx
voice-class sip url
voice-encap
voice-group
voicemail (cm-fallback)
voicemail (telephony-service)
voice-port
voice-port (MGCP profile)
voice-port busyout
voip-incoming translation-profile
voip-incoming translation-rule
volume
web admin customer
web admin system
web customize load
Cisco IOS Voice Commands:
V through W
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 dBm to -62 dBm. Available only when session protocol multicast is configured.
|
Defaults
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 the Cisco 3600 series.
|
12.0(4)T
|
This command was implemented as a dial-peer command on the 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.
On the Cisco MC3810, VAD can also be assigned to the voice port using the vad (voice-port) command. On the Cisco MC3810 multiservice concentrator, if you enable VAD on the dial peer for Voice over Frame Relay switched calls or permanent calls, the dial-peer setting overrides the VAD setting on the voice port.
Note On the Cisco MC3810, the vad (dial-peer) command is enabled by default. The vad (voice-port) command is disabled by default.
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.
|
vad (voice-port)
To enable voice-activity detection (VAD) for the calls using a particular voice port, use the vad command in voice-port configuration mode. To disable VAD, use the no form of this command.
vad
no vad
Syntax Description
This command has no arguments or keywords.
Defaults
VAD is not enabled.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced as a voice-port command on the Cisco MC3810.
|
Usage Guidelines
This command applies to Voice over Frame Relay and Voice over ATM on Cisco MC3810 multiservice concentrators.
Use this command to enable voice activity detection. With VAD, silence is not sent over the network; only audible speech is sent. If you enable VAD, the sound quality is slightly degraded but the connection monopolizes much less bandwidth. If you use the no form of this command, VAD is disabled on the voice port. 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.
Note It is recommended that you use the vad command in dial-peer configuration mode.
Examples
The following example enables VAD:
Related Commands
Command
|
Description
|
comfort-noise
|
Generates background noise to fill silent gaps during calls if VAD is activated.
|
vad (dial peer)
|
Enables VAD for the calls using a particular dial peer.
|
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. Default is 200.
|
Defaults
200 milliseconds
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on the Cisco 2600 series and Cisco 3660.
|
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 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. Default is 4.
|
Defaults
4 milliseconds
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on the Cisco 2600 series and Cisco 3660.
|
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 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.
|
Defaults
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 the Cisco 2600 series and Cisco 3660.
|
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. Default is 100.
|
Defaults
100 milliseconds
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on the Cisco 2600 series and Cisco 3660.
|
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.
|
Defaults
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 the Cisco MC3810.
|
12.1(5)XM
|
This command was implemented on the Cisco 3600 series and modified to support SGCP and MGCP.
|
12.2(2)T
|
This command was integrated into this release.
|
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.
|
Defaults
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 this release.
|
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.
|
vm-device-id (ephone)
To define the voice-mail ID string, use the vm-device-id command in ephone configuration mode. To disable this feature, use the no form of this command.
vm-device-id id-string
no command id-string
Syntax Description
id-string
|
Voice-mail-device port identification (ID) string; for example, CiscoUM-VI1 for the first port and CiscoUM-VI2 for the second port.
|
Defaults
No default behavior or values
Command Modes
Ephone configuration
Command History
Release
|
Modification
|
12.2(2)XT
|
This command was introduced on the Cisco 1750, Cisco 1751, Cisco 2600, Cisco 3600, and Cisco IAD2420.
|
12.2(8)T
|
This command was implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
This command was implemented on the Cisco 2600-XM and Cisco 2691.
|
12.2(11)T
|
This command was implemented on the Cisco 1760.
|
Usage Guidelines
Use this command to define the voice-mail-device ID string. The voice-mail port registers with a device ID instead of a MAC address. To distinguish among different voice-mail ports, voice-mail-device ID is used. The voice-mail-device ID is configured to a Cisco IP phone port, which maps to a corresponding voice-mail port.
Examples
The following example shows how to set the voice-mail device ID to CiscoUM-V11:
Router(config-ephone) vm-device ID CiscoUM-VI1
Related Commands
Command
|
Description
|
voicemail (telephony-service)
|
Configures the telephone number that is speed-dialed when the messages button on a Cisco IP phone is pressed.
|
vm-integration
To enable voice-mail integration with dual-tone multifrequency (DTMF) and analog voice-mail systems and to enter voice-mail integration configuration mode, use the vm-integration command in global configuration mode. To disable voice-mail integration, use the no form of this command.
vm-integration
no vm-integration
Syntax Description
This command has no arguments or keywords.
Defaults
Voice-mail integration is disabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(2)XT
|
For Cisco IOS Telephony Service, this command was introduced on the Cisco 1750, Cisco 1751, Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.
|
12.2(8)T
|
For Cisco IOS Telephony Service, this command was implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
For Cisco IOS Telephony Service,this command was implemented on the Cisco 2600XM and Cisco 2691.
|
12.2(11)T
|
For Cisco IOS Telephony Service, this command was implemented on the Cisco 1760.
|
12.2(13)T
|
This command was implemented on Cisco Survivable Remote Site Telephony, Version 2.02.
|
Usage Guidelines
The vm-integration command allows you to enter voice-mail integration configuration mode and allows integration with DTMF and analog voice-mail systems.
Examples
The following example enters voice-mail integration configuration mode:
Router(config) vm-integration
Router(config-vm-integration)
Related Commands
Command
|
Description
|
pattern direct
|
Configures the DTMF pattern for direct dialing when the user presses the messages button on the phone to access voice-mail messages.
|
pattern ext-to-ext busy
|
Configures the DTMF pattern for forward dialing when an internal extension calls another busy extension and the call is forwarded to a voice-mail system.
|
pattern ext-to-ext no-answer
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension that does not answer and the call is forwarded to voice mail.
|
pattern trunk-to-ext busy
|
Configures the DTMF pattern for forward dialing when an external trunk call reaches a busy extension and the call is forwarded to a voice-mail system.
|
pattern trunk-to-ext no-answer
|
Configures the DTMF pattern for forward dialing when an external trunk call reaches another extension and the call is forwarded to a voice-mail system.
|
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 Before 12.0(7)XK and 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
Cisco-Trunk Permanent Calls to Cisco MC3810 Multiservice Concentrators Running Cisco IOS Releases Before 12.0(7)XK and 12.1(2)T
vofr cisco
no vofr cisco
FRF.11 Trunk Calls
vofr [data cid] [call-control cid]
no vofr [data cid] [call-control cid]
Syntax Description
data cid
|
(Required for Cisco-trunk permanent calls; optional for switched calls) Reserved subchannel for data other than the default subchannel. Range is from 4 to 255. Default is 4.
|
call-control cid
|
(Optional) Reserved subchannel for call-control signaling. Range is from 4 to 255. Default is 5. Not supported on the Cisco MC3810.
|
cisco cid
|
(Optional) Reserved subchannel for Cisco-proprietary voice encapsulation for VoFR. Data is carried on CID 4 and call-control on CID 5. This option is required when configuring switched calls or Cisco trunks to Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T.
If you are configuring switched calls or Cisco trunks to Cisco MC3810 running Cisco IOS Release 12.0(7)XK and 12.1(2)T and later releases, do not use this option.
|
Defaults
Disabled
Command Modes
Frame relay DLCI configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.
|
12.0(4)T
|
This command was integrated into this release.
|
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 this release.
|
Usage Guidelines
Table 159 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 159 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
|
Switched call (user dialed or auto-ringdown) to a Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T
|
vofr cisco2
|
Cisco-trunk permanent call (private-line) to other routers supporting VoFR
|
vofr data cid call-control cid
|
Cisco-trunk permanent call (private-line) to a Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T
|
vofr cisco
|
FRF.11 trunk call (private-line) to other routers supporting VoFR
|
vofr [data cid] [call-control cid]3
|
Usage Restrictions for Cisco IOS Releases Before 12.0(7)XK and 12.1(2)T
This section describes restrictions for using the vofr command in releases before Cisco IOS Release 12.0(7)XK and 12.1(2)T. Beginning in Cisco IOS Release 12.0(7)XK and 12.1(2)T, these restrictions no longer apply.
When you use the vofr command without the cisco option, all subchannels on the DLCI are configured for FRF.11 encapsulation. If you enter the vofr command without any keywords or arguments, the data subchannel is CID 4 and there is no call-control subchannel.
Table 160 describes special conditions and restrictions for the use of the vofr command on the Cisco MC3810 running releases before 12.0(7)XK and 12.1(2)T.
Table 160 Using the vofr Command with the Cisco MC3810
Type of Call
|
Conditions and Restrictions
|
FRF.11 trunks
|
1. Do not use the cisco option or the call-control option.
2. Use vofr or vofr data cid.
|
Cisco trunks
|
1. Must use vofr cisco.
|
switched-vofr
|
1. Must use vofr cisco.
|
If you select the "data" option, enter a numeric value to complete the command. If you select the call-control option, you do not enter a numeric value if you wish to accept the default call-control subchannel. See the following examples for clarification.
When you use the vofr command on a Cisco MC3810 multiservice concentrator without the "cisco" option, switched calls are not permitted. You can make only permanent FRF.11-trunk calls.
Note It is not possible to configure the call-control option on a Cisco MC3810. If you configure this option, the setting is ignored.
Examples
The following example, beginning in global configuration mode, shows how to enable VoFR on serial interface 1/1, DLCI 100 on a Cisco 2600 series, Cisco 3600 series, or Cisco 7200 series router or on a Cisco MC3810. 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:
To configure a Cisco router or Cisco MC3810 for a VoFR application with an older release of the Cisco MC3810 (before Release 12.0(3)XG), 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.
Defaults
Disabled
Command Modes
Service profile configuration mode
Command History
Release
|
Modification
|
12.2(2)XA
|
This command was introduced on the Cisco AS5350 and Cisco AS5400.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(11)T
|
This command was integrated into this release.
|
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. Range is from 1 to 3600. Default is 10. This value cannot be set higher than the time configured for the capacity update interval.
|
Defaults
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 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
Syntax Description
carrier
|
Carrier-code address family.
|
trunk-group
|
Trunk-group address family.
|
prefix
|
E.164 prefix.
|
seconds
|
Transition time, in seconds. Range is from 0 to 60. Default is 10.
|
Defaults
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 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.
|
Defaults
Capacity reporting 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 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
|
Interval, in seconds. Range is from 10 to 3600. Default is 25.
|
Defaults
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.
|
Defaults
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, 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
|
Number of data-points. Range is from 10 to 50. Default is 30.
|
Defaults
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 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
|
Recording interval, in minutes. Range is from 10 to 1000. Default is 60.
|
Defaults
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
|
Reporting interval, in seconds. Range is from 10 to 10000. Default is 25.
|
Defaults
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 you use the no version of this command 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.
|
Defaults
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: Cisco 2600 series, Cisco 3620, Cisco 3640, Cisco 3660 series, Cisco AS5350, Cisco AS5400, Cisco AS5850, Cisco AS5300, Cisco AS5800, and Cisco MC3810.
|
12.2(15)T
|
The header-only argument was removed and the short-header argument was added.
|
Usage Guidelines
The user can control the contents of the standardized header. The 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 are 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 the 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 161 describes the significant fields shown in the display.
Table 161 voice call debug full-guid Field Descriptions
Field
|
Description
|
VTSP:(0:D):0:0:4385
|
Identifies the VTSP module, port name, channel number, DSP slot, and DSP channel number.
|
vtsp_insert_cdb
|
Identifies the function name.
|
CCAPI
|
Identifies the CCAPI module.
|
The following is sample output for the voice call debug command 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 output "//-1/" 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 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.
Defaults
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 this release.
|
12.2(1)
|
This command was integrated into this release.
|
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 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
|
High-watermark value, as a percentage. Range is from 50 to 100. Default is 80. If set to 100, this trigger turns off.
|
Defaults
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
|
Low-watermark value, as a percentage. Range is from 0 to 30. Default is 10. If set to 0, this trigger turns off.
|
Defaults
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 percentage change 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
|
Percentage change. Range is from 0 to 100. Default is 30. If set to 0, this trigger turns off.
|
Defaults
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
|
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
|
Voice-class AAA identifier. Range is from 1 to 10000. There is no default.
|
Defaults
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.
|
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
|
Description
|
authentication method
|
Specifies an authentication method for calls coming into the defined dial peer.
|
authorization method
|
Specifies an authorization method for calls coming into the defined dial peer.
|
method
|
Specifies an accounting method for calls coming into the defined dial peer.
|
accounting suppress
|
Disables accounting that is automatically generated by the service provider module for a specific dial peer.
|
voice-class aaa
|
Applies properties defined in the voice class to a specific dial peer.
|
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
|
Voice-class AAA identifier. Range is from 1 to 10000. There is no default.
|
Defaults
No default behaviors or values
Command Modes
Dial peer 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.
|
Usage Guidelines
Properties that are configured in voice class AAA configuration mode can be applied to a dial peer by using the voice-class aaa command in dial peer configuration mode.
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 the voice-class aaa command in dial peer configuration mode.
incoming called-number 50..
session target ipv4:1.5.31.201
Related Commands
Command
|
Description
|
voice class aaa
|
Enables dial-peer-based VoIP AAA configurations.
|
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
|
Unique identifier assigned to one voice class. Range is from 1 to 10000. There is no default.
|
Defaults
No voice class is configured for busyout functions.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600, Cisco 3600, and Cisco MC3810.
|
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
|
Description
|
busyout monitor ethernet
|
Configures a voice port to monitor a local Ethernet interface for events that would trigger a voice-port busyout.
|
busyout monitor probe
|
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.
|
busyout monitor serial
|
Configures a voice port to monitor a serial interface for events that would trigger a voice-port busyout.
|
show voice busyout
|
Displays information about the voice busyout state.
|
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
|
Unique identifier assigned to the voice class. Range is from 1 to 10000. There is no default.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(2)XH
|
This command was introduced on the Cisco AS5300.
|
12.0(7)T
|
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into this release.
|
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 Voice over IP (VoIP) dial peer.
Note The voice class codec command in global configuration mode is entered without the hyphen. The voice-class codec command in dial-peer configuration mode is entered with the 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
|
Description
|
codec preference
|
Specifies a list of preferred codecs to use on a dial peer.
|
test voice port detector
|
Defines the order of preference in which network dial peers select codecs.
|
voice-class codec (dial peer)
|
Assigns a previously configured codec selection preference list to a dial peer.
|
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
|
Unique identifier 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.
|
Defaults
Dial peers have no codec voice class assigned.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(2)XH
|
This command was introduced on the Cisco AS5300.
|
12.0(7)T
|
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into this release.
|
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
|
Description
|
show dial-peer voice
|
Displays the configuration for all dial peers configured on the router.
|
test voice port detector
|
Defines the order of preference in which network dial peers select codecs.
|
voice class codec
|
Enters voice-class configuration mode and assigns an identification tag number for a codec voice class.
|
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
|
Descriptive identifier for this class of custom call-progress tones that associates this set of custom call-progress tones with voice ports.
|
Defaults
No voice class of custom call-progress tones is created.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(5)XM
|
This command was introduced on the Cisco 2600, Cisco 3600, and Cisco MC3810.
|
12.2(2)T
|
This command was implemented on the Cisco 1750.
|
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
|
Description
|
dualtone
|
Defines the tone and cadence for a custom call-progress tone.
|
supervisory custom-cptone
|
Associates a class of custom call-progress tones with a voice port.
|
voice class dualtone-detect-params
|
Modifies the boundaries and limits for call-progress tones.
|
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
|
Unique identifier assigned to one voice class. Range is from 1 to 10000. There is no default.
|
Defaults
No voice class is configured for tone detection parameters.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(3)T
|
This command was introduced on the Cisco 2600 series, Cisco 3600, and Cisco MC3810.
|
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
|
Description
|
supervisory disconnect dualtone voice-class
|
Assigns a previously configured voice class for FXO supervisory disconnect tone to a voice port.
|
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
|
Unique identifier assigned to a voice class. Range is from 1 to 10000. There is no default.
|
Defaults
No voice class is configured for defining answer-supervision tolerance limits.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(5)XM
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.2(2)T
|
This command was implemented on the Cisco 1750.
|
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
|
Description
|
supervisory dualtone-detect-params
|
Assigns the boundary and detection tolerance parameters defined by the voice class dualtone-detect-params command to a voice port.
|
voice class custom-cptone
|
Creates a voice class for defining custom call-progress tones.
|
voice class h323
To create an H.323 voice class that is independent of a dial peer and can be used on multiple dial peers, use the voice class h323 command in global configuration mode. To remove the voice class, use the no form of this command.
voice class h323 tag
no voice class h323
Syntax Description
tag
|
Unique identifier assigned to the voice class. Range is from 1 to 10000. There is no default.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(2)T
|
This command was introduced on the Cisco 1700, Cisco 2600 series, Cisco 3600 series, Cisco 7200, Cisco AS5300, Cisco uBR910, and Cisco uBR924.
|
Usage Guidelines
The voice class h323 command in global configuration mode does not include a hyphen. The voice-class h323 command in dial-peer configuration mode includes a hyphen.
Examples
The following example creates an H.323 voice class labeled 1:
Related Commands
Command
|
Description
|
h225 timeout tcp establish
|
Sets the H.225 TCP timeout value.
|
voice-class h323 (dial peer)
To assign an H.323 voice class to a VoIP dial peer, use the voice-class h323 command in dial-peer configuration mode. To remove the voice class from the dial peer, use the no form of this command.
voice-class h323 tag
no voice-class h323 tag
Syntax Description
tag
|
Unique identifier assigned to the voice class. Range is from 1 to 10000. There is no default.
|
Defaults
The dial peer does not use an H.323 voice class.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.1(2)T
|
This command was introduced.
|
Usage Guidelines
The voice class that you assign to the dial peer must be configured using the voice class h323 in global configuration mode.
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.
The voice-class h323 command in dial-peer configuration mode includes a hyphen and in global configuration mode does not include a hyphen.
Examples
The following example shows how to create an H.323 voice class and then assign it to a dial peer:
Related Commands
Command
|
Description
|
show dial-peer voice
|
Displays the configuration for all dial peers configured on the router.
|
voice class h323
|
Enters voice-class configuration mode and assigns an identification tag number for an H.323 voice class.
|
voice class permanent
To create a voice class for a Cisco trunk or FRF.11 trunk, use the voice class permanent command in global configuration mode. To delete the voice class, use the no form of this command.
voice class permanent tag
no voice class permanent tag
Syntax Description
tag
|
Unique identifier assigned to the voice class. Range is from 1 to 10000. There is no default.
|
Defaults
No voice class is configured.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810.
|
12.0(4)T
|
This command was integrated into this release.
|
12.1(3)T
|
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
|
Usage Guidelines
The voice class permanent command can be used for Voice over Frame Relay (VoFR), Voice over ATM (VoATM), and Voice over IP (VoIP) trunks.
The voice class permanent command in global configuration mode is entered without a hyphen. The voice-class permanent command in dial-peer and voice-port configuration modes is entered with a hyphen.
Examples
The following example shows how to create a permanent voice class starting from global configuration mode:
Related Commands
Command
|
Description
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type for a network dial peer.
|
voice-class permanent
|
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a network dial peer.
|
voice-class permanent (dial-peer)
To assign a previously configured voice class for a Cisco trunk or FRF.11 trunk to a network dial peer, use the voice-class permanent command in dial-peer configuration mode. To remove the voice-class assignment from the network dial peer, use the no form of this command.
voice-class permanent tag
no voice-class permanent tag
Syntax Description
tag
|
Unique identifier assigned to the voice class. The tag number maps to the tag number created using the voice class permanent global configuration command. Range is from 1 to 10000. There is no default.
|
Defaults
Network dial peers have no voice class assigned.
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810.
|
12.0(4)T
|
This command was integrated into this release.
|
12.1(3)T
|
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
|
Usage Guidelines
You can assign one voice class to any given network dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
You cannot assign a voice class to a plain old telephone service (POTS) dial peer.
The voice-class permanent command in dial-peer configuration mode is entered with a hyphen. The voice class permanent command in global configuration mode is entered without a hyphen.
Examples
The following example assigns a previously configured voice class to a Voice over Frame Relay (VoFR) network dial peer:
Related Commands
Command
|
Description
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type for a network dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice-class permanent (voice-port)
To assign a previously configured voice class for a Cisco trunk or FRF.11 trunk to a voice port, use the voice-class permanent command in voice-port configuration mode. To remove the voice-class assignment from the voice port, use the no form of this command.
voice-class permanent tag
no voice-class permanent tag
Syntax Description
tag
|
Unique identifier assigned to the voice class. The tag number maps to the tag number created using the voice class permanent global configuration command. Range is 1 to 10000. There is no default.
|
Defaults
Voice ports have no voice class assigned.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
12.0(3)XG
|
This command was introduced on the Cisco MC3810.
|
12.0(4)T
|
This command was integrated into this release.
|
12.1(3)T
|
This command was implemented as a voice-port configuration command on the Cisco 2600 series and Cisco 3600 series.
|
Usage Guidelines
You can assign one voice class to any given voice port. If you assign another voice class to a voice port, the last voice class assigned replaces the previous voice class.
The voice-class permanent command in voice-port configuration mode is entered with a hyphen. The voice class permanent command in global configuration mode is entered without a hyphen.
Examples
The following example assigns a previously configured voice class to voice port 1/1 in a Cisco MC3810 multiservice concentrator:
The following example assigns a previously configured voice class to voice port 1/1/0 in a Cisco 3600 series router:
Related Commands
Command
|
Description
|
signal keepalive
|
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
|
signal pattern
|
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
|
signal timing idle suppress-voice
|
Configures the signal timing parameter for the idle state of a call.
|
signal timing oos
|
Configures the signal timing parameter for the OOS state of a call.
|
signal-type
|
Sets the signaling type for a network dial peer.
|
voice class permanent
|
Creates a voice class for a Cisco trunk or FRF.11 trunk.
|
voice confirmation-tone
To disable the two-beep confirmation tone for private line, automatic ringdown (PLAR), or PLAR off-premises extension (OPX) connections, use the voice confirmation-tone command in voice-port configuration mode. To enable the two-beep confirmation tone, use the no form of this command.
voice confirmation-tone
no voice confirmation-tone
Syntax Description
This command has no arguments or keywords.
Defaults
The two-beep confirmation tone is heard on PLAR and PLAR OPX connections.
Command Modes
Voice-port configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810.
|
Usage Guidelines
This command applies only to the Cisco MC3810 multiservice concentrator.
Use this command to disable the two-beep confirmation tone that a caller hears when picking up the handset for PLAR and PLAR OPX connections. This command is valid only if the voice-port connection command is set to PLAR or PLAR OPX.
Examples
The following example disables the two-beep confirmation tone on voice port 1/1 on the Cisco MC3810 multiservice concentrator:
Related Commands
Command
|
Description
|
connection
|
Specifies a connection mode for a voice port.
|
voice dnis-map
To create or modify a Digital Number Identification Service (DNIS) map, use the voice dnis-map command in global configuration mode. To delete a DNIS map, use the no form of this command.
voice dnis-map map-name [url]
no voice dnis-map map-name
Syntax Description
map-name
|
Name of the DNIS map.
|
url
|
(Optional) URL of an external text file that contains a list of DNIS entries.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(2)XB
|
This command was introduced on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
|
12.2(11)T
|
This command was implemented on the Cisco 3640 and Cisco 3660.
|
Usage Guidelines
A DNIS map is a table of DNIS numbers associated with a single dial peer. For applications such as VoiceXML, using a DNIS map makes it possible to configure a single dial peer for all DNIS numbers used to refer to VoiceXML documents. Keep the following considerations in mind when using voice DNIS maps.
•A separate entry must be made for each DNIS entry in a DNIS map. Wildcards are not supported.
•If a URL is not supplied, the command enters DNIS-map configuration mode, permitting the entry of DNIS numbers by using the dnis command.
•The URL argument points to the location of an external text file containing a list of DNIS entries (for example: tftp://dnismap.txt). This allows the administrator to maintain a single master file of all DNIS map entries, if desired, rather than configuring the DNIS entries on each gateway.
The name of the text file extension is not significant; .doc, .txt, or .cfg are all acceptable because the extension is not checked. The entries in the file should look the same as a DNIS entry configured in Cisco IOS software (for example: dnis 5553305 url tftp://global/tickets/movies.vxml).
•External text files used for DNIS maps must be stored on TFTP servers; they cannot be stored on HTTP servers.
•To associate a DNIS map with a dial peer, use the dnis-map command.
•To view the configuration information for DNIS maps, use the show voice dnis-map command.
Examples
The following example shows how the voice dnis-map command is used to create a DNIS map:
The following example shows the voice dnis-map command used with a URL that specifies the location of a text file containing the DNIS entries:
voice dnis-map dmap2 tftp://keyer/dmap2/dmap2.txt
Following is an example of the contents of a text file comprising a DNIS map:
!Example dnis-map with 8 entries.
dnis 5551212 url tftp://global/ticket/vapptest1.vxml
dnis 5551111 url tftp://global/ticket/vapptest2.vxml
dnis 5551234 url tftp://global/ticket/vapptest3.vxml
Related Commands
Command
|
Description
|
dnis
|
Adds a DNIS number to a DNIS map.
|
dnis-map
|
Associates a DNIS map with a dial peer.
|
show voice dnis-map
|
Displays configuration information about DNIS maps.
|
voice dnis-map load
|
Reloads a DNIS map that has changed since the previous load.
|
voice dnis-map load
To reload a DNIS map that has been modified, use the voice dnis-map load command in privileged EXEC mode.
voice dnis-map load map-name
Syntax Description
map-name
|
Name of the DNIS map to reload.
|
Defaults
No default behavior or values
Command Modes
Privileged EXEC
Command History
Release
|
Modification
|
12.2(2)XB
|
This command was introduced on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
|
12.2(11)T
|
This command was implemented on the Cisco 3640 and Cisco 3660.
|
Usage Guidelines
This command reloads a DNIS map residing on an external server. Use this command when the DNIS map file has changed since the previous load.
To create or modify a DNIS map, use the voice dnis-map command.
Examples
The following example shows how the voice dnis-map load command is used to reload a DNIS map named "mapfile1":
Router# voice dnis-map load mapfile1
Related Commands
Command
|
Description
|
dnis
|
Adds a DNIS number to a DNIS map.
|
dnis-map
|
Associates a DNIS map with a dial peer.
|
show voice dnis-map
|
Displays configuration information about DNIS maps.
|
voice dnis-map
|
Enters DNIS map configuration mode to create a DNIS map.
|
voice echo-canceller extended
To enable the G.168 extended echo canceller (EC) on the Cisco 1700 series or Cisco ICS7750, use the voice echo-canceller extended command in global configuration mode. To return to the Cisco-proprietary G.165 default EC, use the no form of this command.
voice echo-canceller extended
no voice echo-canceller extended
Syntax Description
This command has no arguments or keywords.
Defaults
The G.168 extended EC is not enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(13)T
|
This command was introduced.
|
Usage Guidelines
You do not have to shut down all the voice ports on the Cisco 1700 series or Cisco ICS7750 in order to switch the echo canceller, but you should make sure that when you switch the echo canceller, there are no active calls on the router.
Because echo cancellation is an invasive process that can minimally degrade voice quality, this command should be disabled if it is not needed.
Note This command is valid only when the echo-canceller coverage command has been configured.
Examples
To switch to the G.168 extended EC from the Cisco default EC on the Cisco 1700 series or Cisco ICS7750 platforms, use the following command in global configuration mode:
Router(config)# voice echo-canceller extended
Related Commands
Command
|
Description
|
echo-cancel enable
|
Enables the cancellation of voice that is sent and received on the same interface.
|
echo-canceller coverage
|
Adjusts the size of the EC and selects the extended EC when the Cisco default EC is present.
|
voice enum-match-table
To create an ENUM match table for voice calls, use the voice enum-match-table in global configuration mode. To delete the ENUM match table, use the no form of this command.
voice enum-match-table table-number
no voice enum-match-table table-number
Syntax Description
table-number
|
Number of the ENUM match table. Range is from 1 to 15. There is no default.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
This command was introduced.
|
Usage Guidelines
The ENUM match table is a set of rules for matching incoming calls. When a call comes in, its called number is matched against the match pattern of the rule with the highest preference.
If it matches, the replacement pattern is applied to the number. The resulting number and the domain name of the rule are used to make an ENUM query.
If the called number does not match the match pattern, the next rule in order of preference is selected.
Examples
The following example creates ENUM match table 3 for voice calls:
Router(config)# voice enum-match-table 3
Router(config-enum)# rule 1 5/(.*)/ /\1/e164.cisco.com
Router(config-enum)# rule 2 4/^9011\(.*\)/ /\1/e164.arpa
In this table, rule 1 matches any number. The resulting number is the same as the called number. That number and the domain name "e164.cisco.com" are used to make an ENUM query.
Rule 2 matches any number that starts with 9011. The 9011 is removed from the incoming number. The resulting number and the domain name "e164.arpa" are used for the ENUM query.
Suppose an incoming call has a called number of 4085551212. [Rule 2 is applied] first because it has a higher preference. The first few digits, 4085, do not match the 9011 pattern of rule 2, so [rule 1 is applied] next. The called number matches rule 1, and the resulting number is 4085551212. This number and "e164.cisco.com" form the ENUM query (2.1.2.1.5.5.5.8.0.4.e164.cisco.com).
Related Commands
Command
|
Description
|
rule (ENUM configuration)
|
Defines the matching, replacement, and rejection patterns for an ENUM match table.
|
show voice enum-match-table
|
Displays the configuration of voice ENUM match tables.
|
test enum
|
Tests the functionality of an ENUM match table.
|
voice hpi capture
To allocate the Host Port Interface (HPI) capture buffer size (in bytes) and to set up or change the destination URL for captured data, use the voice hpi capture command in global configuration mode. To stop all logging and file operations, to disable data transport from the capture buffer, and to automatically set the buffer size to 0, use the no form of this command.
voice hpi capture [buffer size | destination url]
no voice hpi capture buffer size
Syntax Description
buffer size
|
(Optional) Size of the HPI capture buffer, in bytes. Range is from 328 to 9000000. Default is 328.
|
destination url
|
(Optional) Destination URL for storing captured data.
|
Defaults
328 bytes (no buffer is used if it is not configured explicitly)
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(10)
|
This command was introduced.
|
12.2(11)T
|
This command was integrated into this release.
|
Usage Guidelines
If you want to change the size of an existing non-zero buffer, you must first reset it to 0 and then change it from 0 to the new size.
The destination url option sets up or changes the destination URL for captured data. To disable data transport from the capture buffer, use the no form of the command. If the buffer is allocated, captured data is sent to the current URL (if it was already configured) until the new URL is specified.
If a new URL differs from the current URL and logging is enabled, the current URL is closed and all further data is sent to the new URL. Entering a blank URL or prefixing the command with no disables data transport from the capture buffer, and (if capture is enabled) captured data is stored in the capture buffer until it reaches its capacity.
Once the buffer-queueing program is running, the transport process attempts to connect to a new or existing "capture destination" URL. A version message is written to the URL, and if the message is successfully received, any further messages placed into the message queue are written to that URL. If a new URL is entered using the voice hpi capture destination url command, the open URL is closed, and the system attempts to write to the new URL. If the new URL does not work, the transport process exits. The transport process is restarted when another URL is entered or the system is restarted.
The buffer size option sets the maximum amount of memory (in bytes) that the capture system allocates for its buffers when it is active. The capture buffer is where the captured messages are stored before they are sent to the URL specified by the capture destination. The system is started by choosing the amount of memory (greater than 0 bytes) that the buffer-queueing system can allocate to the free message pool. HPI messages can then be captured until buffer capacity is reached. Entering 0 for the buffer size and prefixing the command with no stops all logging and file operations and automatically sets the buffer size to 0.
The voice hpi capture command can be saved with the router configuration so that the command is active during router startup. This allows you to capture the HPI messages sent during router bootup before the CLI is enabled. After you have configured the buffer size in the running configuration (valid range is from 328 to 9000000), save it to the startup configuration using the write command or to the TFTP server using the copy run tftp command.
Caution Using the message logger feature in a production network environment impacts CPU and memory usage on the gateway.
Examples
The following example changes the size (in bytes) of the HPI capture buffer and initializes the buffer-queueing program:
Router# configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)# voice hpi capture buffer 40000
03:23:31:caplog:caplog_cli_interface:hpi capture buffer size set to 40000 bytes
03:23:31:caplog:caplog_logger_init:TRUE, Started task HPI Logger (PID 64)
03:23:31:caplog:caplog_cache_init:TRUE, malloc_named(39852), 123 elements (each 324 bytes
big)
03:23:31:caplog:caplog_logger_proc:Attempting to open ftp://172.23.184.233/c:b-38-117
03:23:32:%SYS-5-CONFIG_I:Configured from console by console
The following example sets the capture destination by entering a destination URL using FTP:
Router# configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Router(config)# voice hpi capture destination ftp://172.23.184.233/c:b-38-117a
04:05:10:caplog:caplog_cli_interface:hpi capture
destination:ftp://172.23.184.233/c:b-38-117a
04:05:10:caplog:caplog_logger_init:TRUE, Started task HPI Logger (PID 19)
04:05:10:caplog:caplog_cache_init:Cache must be at least 324 bytes
04:05:10:caplog:caplog_logger_proc:Terminating...
Related Commands
Command
|
Description
|
debug hpi
|
Turns on the debug output for the logger.
|
show voice hpi capture
|
Displays the capture status and statistics.
|
voice hunt
To configure an originating or tandem router so that it continues dial-peer hunting if it receives a user-busy disconnect code from a destination router, use the voice hunt command in global configuration mode. To configure the router so that it stops dial-peer hunting if it receives a user-busy disconnect code (the default option), use the no form of this command.
voice hunt {user-busy | invalid-number | unassigned-number}
no voice {user-busy | invalid-number | unassigned-number}
Syntax Description
user-busy
|
Router continues dial-peer hunting if it receives a user-busy disconnect cause code from a destination router.
|
invalid-number
|
Router stops dial-peer hunting if it receives a an invalid-number disconnect cause code from a destination router.
|
unassigned-number
|
Router stops dial-peer hunting if it receives an unassigned-number disconnect cause code from a destination router.
|
Defaults
The default depends on the disconnect cause code. By default, the router stops dial-peer hunting if it receives the user-busy disconnect cause code. By default, the router continues dial-peer hunting if it receives an invalid-number, or an unassigned-number disconnect cause code.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced for VoFR on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810. It was also introduced for VoIP on Cisco 2600 series and Cisco 3600 series.
|
12.0(7)T
|
This command was implemented for VoIP on the Cisco AS5300 and Cisco AS5800.
|
12.0(7)XK
|
This command was implemented for VoIP on the Cisco MC3810.
|
12.1(2)T
|
This command was implemented for VoIP on the Cisco MC3810.
|
12.1(3)XI
|
The invalid-number and unassigned-number keywords were added, and the command name was changed to voice hunt.
|
12.1(5)T
|
This command was integrated into this release.
|
Usage Guidelines
This command applies to routers that act as originating or tandem nodes in a Voice over IP, Voice over Frame Relay, or Voice over ATM environment.
This command is used for a configuration in which an originating or tandem router is configured with multiple dial peer entries that route a call to the same destination number, but on different destination routers. In this configuration, after all routes to the first router entry in the dial-peer list are active, a new call does not "roll over" to the next router in the dial-peer list.
This failure to route to the second destination router happens when the bandwidth on the voice interface is greater than the maximum capacity of the first destination router. This condition allows the originating or tandem router to attempt to place a new call to the first destination router because it has indications from the first destination router that there is more capacity based on the bandwidth setting. When the first destination router receives the call, if all of the ports are in use, the destination router returns a "user-busy" disconnect reason code to the originating or tandem router.
The originating or tandem router interprets the disconnect reason code as "unavailable destination" for the call and returns a busy tone to the initiating caller.
The originating or tandem router fails to try other routers in the dial-peer list after receiving a "user disconnect" reason code, and so it terminates the call attempt. By using this command, you can perform dial-peer hunting on multiple destination routers even if the originating or tandem router receives a "user-busy" disconnect reason code from one of the destination routers.
Examples
The following example configures the originating or tandem router to continue dial-peer hunting if it receives a "user-busy" disconnect code from a destination router:
The following example configures the originating or tandem router to continue dial-peer hunting if it receives an "invalid-number" disconnect code from a destination router:
voice hunt invalid-number
Related Commands
Command
|
Description
|
huntstop
|
Disables all further dial-peer hunting if a call fails when using hunt groups.
|
preference
|
Indicates the preferred order of a dial peer within a rotary hunt group.
|
voice local-bypass
To configure local calls to bypass the digital signal processor (DSP), use the voice local-bypass command in global configuration mode. To direct local calls through the DSP, use the no form of this command.
voice local-bypass
no voice local-bypass
Syntax Description
This command has no arguments or keywords.
Defaults
Local calls bypass the DSP.
Command Modes
Global configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced.
|
12.0(7)XK
|
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.1(2)T
|
This command was integrated into this release.
|
Usage Guidelines
Local calls (calls between voice ports on a router or concentrator) normally bypass the DSP to minimize use of system resources. Use the no form of the voice local-bypass command if you need to direct local calls through the DSP. Input gain and output attenuation can be configured only if calls are directed through the DSP.
Examples
The following example configures a Cisco MC3810 multiservice concentrator or Cisco 2600 series or Cisco 3600 series router to pass local calls through the DSP:
Related Commands
Command
|
Description
|
input gain
|
Configures a specific input gain value.
|
output attenuation
|
Configures a specific output attenuation value.
|
voice rtp send-recv
To establish a two-way voice path when the Real-Time Transport Protocol (RTP) channel is opened, use the voice rtp send-recv command in global configuration mode. To reset to the default, use the no form of this command.
voice rtp send-recv
no voice rtp send-recv
Syntax Description
This command has no arguments or keywords.
Defaults
The voice path is cut-through in only the backward direction when the RTP channel is opened.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(5)T
|
This command was introduced on the Cisco 2600, Cisco 3600, Cisco 7200, Cisco 7500, Cisco AS5300, Cisco AS5800, and Cisco MC3810.
|
12.2(2)XA
|
This command was implemented on the Cisco AS5350 and Cisco AS5400.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(11)T
|
This command was integrated into this release.
|
Usage Guidelines
This command should be enabled only when the voice path must be cut-through (established) in both the backward and forward directions before a Connect message is received from the destination switch. This command affects all VoIP calls when it is enabled.
Examples
The following example enables the voice path to cut-through in both directions when the RTP channel is opened:
voice service
To enter voice-service configuration mode and to specify a voice-encapsulation type, use the voice service command in global configuration mode.
voice service {pots | voatm | vofr | voip}
Syntax Description
pots
|
Telephony voice service.
|
voatm
|
Voice over ATM (VoATM) encapsulation.
|
vofr
|
Voice over Frame Relay (VoFR) encapsulation.
|
voip
|
Voice over IP (VoIP) encapsulation.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into this release.
|
12.1(3)T
|
This command was implemented for VoIP on the Cisco 2600 series and Cisco 3600 series.
|
12.1(3)XI
|
This command was implemented on the Cisco AS5300.
|
12.1(5)T
|
This command was integrated into this release.
|
12.1(5)XM
|
This command was implemented on the Cisco AS5800.
|
12.1(5)XM2
|
This command was implemented on the Cisco AS5350 and Cisco AS5400.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(2)T
|
This command was implemented on the Cisco 7200 series.
|
12.2(11)T
|
This command was implemented on the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850.
|
Usage Guidelines
Voice-service configuration mode is used for packet telephony service commands that affect the gateway globally.
Examples
The following example enters voice-service configuration mode for VoATM service commands:
voice source-group
To define a source IP group for voice calls, use the voice source-group command in global configuration mode. To delete the source IP group, use the no form of this command.
voice source-group name
no voice source-group name
Syntax Description
name
|
Name of the IP group. Maximum length is 31 alphanumeric characters.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
This command was introduced.
|
Usage Guidelines
Use the voice source-group command to assign a name to a set of source IP group characteristics. The terminating gateway uses these characteristics to identify and translate the incoming VoIP call.
Carrier IDs and trunk group labels must not have the same names.
Do not mix carrier IDs and trunk group labels within a source IP group.
A terminating gateway can be configured with carrier ID source IP groups and trunk-group-label source IP groups. The name of the source IP group must be unique to the gateway.
Examples
The following example initiates source IP group "utah2" for VoIP calls:
Router(config)# voice source-group utah2
Related Commands
Command
|
Description
|
access-list
|
Defines a list of source groups for identifying incoming calls.
|
carrier-id (voice source group)
|
Specifies the carrier handling a VoIP call.
|
description (voice source group)
|
Assigns a disconnect cause to a source IP group.
|
h323zone-id (voice source group)
|
Assigns a zone ID to an incoming H.323 call.
|
translation-profile (source group)
|
Assigns a translation profile to a source IP group.
|
trunk-group-label (voice source group)
|
Specifies the trunk handling a VoIP call.
|
voice translation-profile
To define a translation profile for voice calls, use the voice translation-profile command in global configuration mode. To delete the translation profile, use the no form of this command.
voice translation-profile name
no voice translation-profile name
Syntax Description
name
|
Name of the translation profile. Maximum length is 31 alphanumeric characters.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
This command was introduced.
|
Usage Guidelines
After translation rules are defined, they are grouped into profiles. The profiles collect a set of rules that, taken together, translate the called, calling, and redirected numbers in specific ways. Up to 1000 profiles can be defined. Each profile must have a unique name.
These profiles are referenced by trunk groups, dial peers, source IP groups, voice ports, and interfaces for handling call translations.
Examples
The following example initiates translation profile "westcoast" for voice calls. The profile uses translation rules 1, 2, and 3 for various types of calls.
Router(config)# voice translation-profile westcoast
Router(cfg-translation-profile)# translate calling 2
Router(cfg-translation-profile)# translate called 1
Router(cfg-translation-profile)# translate redirect-called 3
Related Commands
Command
|
Description
|
rule (voice translation-rule)
|
Defines call translation criteria.
|
show voice translation-profile
|
Displays one or more translation profiles.
|
translate (translation profiles)
|
Associates a translation rule with a voice translation profile.
|
voice translation-rule
To define a translation rule for voice calls, use the voice translation-rule command in global configuration mode. To delete the translation rule, use the no form of this command.
voice translation-rule number
no voice translation-rule number
Syntax Description
number
|
Unique identifier for the translation rule. Range is from1 to 2147483647. There is no default.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
This command was introduced.
|
Usage Guidelines
Use the voice translation-rule command to create the definition of a translation rule. Each definition includes up to 15 rules that include SED-like expressions for processing the call translation. A maximum of 128 translation rules are supported.
These translation rules are grouped into profiles that are referenced by trunk groups, dial peers, source IP groups, voice ports, and interfaces.
Examples
The following example initiates translation rule 150, Which includes two rules:
Router(config)# voice translation-rule 150
Router(cfg-translation-rule)# rule 1 reject /^408\(.(\)/
Router(cfg-translation-rule)# rule 2 /\(^...\)853\(...\)/ /\1525\2/
Related Commands
Command
|
Description
|
rule (voice translation-rule)
|
Defines the matching, replacement, and rejection patterns for a translation rule.
|
show voice translation-rule
|
Displays the configuration of a translation rule.
|
voice vad-time
To change the minimum silence detection time for voice activity detection (VAD), use the voice vad-time command in global configuration mode. To reset to the default, use the no form of this command.
voice vad-time milliseconds
no voice vad-time
Syntax Description
milliseconds
|
Waiting period, in milliseconds, before silence detection and suppression of voice-packet transmission. Range is from 250 to 65536. Default is 250.
|
Defaults
250 milliseconds
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.1(2)T
|
This command was integrated into this release.
|
Usage Guidelines
This command affects all voice ports on a router or concentrator, but it does not affect calls already in progress.
You can use this command in transparent common-channel signaling (CCS) applications in which you want VAD to activate when the voice channel is idle, but not during active calls. With a longer silence detection delay, VAD reacts to the silence of an idle voice channel, but not to pauses in conversation.
This command does not affect voice codecs that have ITU-standardized built-in VAD features—for example, G.729B, G.729AB, G.723.1A. The VAD behavior and parameters of these codecs are defined exclusively by the applicable ITU standard.
Examples
The following example configures a 20-second delay before VAD silence detection is enabled:
Related Commands
Command
|
Description
|
vad (dial peer)
|
Enables voice activity detection on a network dial peer.
|
voice-card
To enter the voice-card configuration mode and configure a voice card, use the voice-card command in global configuration mode.
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.
–1 is the network module slot .
•Cisco 3600 series
–1to 6 are network-module slots.
•Cisco 3660
–7 is AIM slot 0.
–8 is AIM slot 1.
•Cisco MC3810 with one or two high-performance voice-compression modules (HCMs) installed
–0 applies to the entire chassis.
|
Defaults
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 this release.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into this release.
|
12.2(2)XB
|
Values for the slot argument were updated to include AIMs.
|
12.2(8)T
|
This command was integrated into this release.
|
12.2(13)T
|
This command was 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 this release.
|
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, and on high-performance compression modules on Cisco MC3810 multiservice access concentrators.
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.
•On the Cisco MC3810, the slot argument is always 0, and the changes that are made in voice-card mode apply to the entire Cisco MC3810. On the Cisco MC3810, the voice-card command is available only if the chassis is equipped with one or two HCMs.
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 on a Cisco 2600 series or Cisco 3600 series router:
The following example enters voice-card configuration mode on a Cisco MC3810:
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 sip rel1xx
To enable all Session Initiation Protocol (SIP) provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint, use the voice-class sip rel1xx command in dial-peer configuration mode. To reset to the default, use the no form of this command.
voice-class sip rel1xx {supported value | require value | system | disable}
no sip rel1xx
Syntax Description
supported value
|
Supports reliable provisional responses. The value argument may have any value, as long as both the user-agent client (UAC) and user-agent server (UAS) configure it the same.
|
require value
|
Requires reliable provisional responses. The value argument may have any value, as long as both the UAC and UAS configure it the same.
|
system
|
Uses the value configured in voice service mode. This is the default.
|
disable
|
Disables the use of reliable provisional responses.
|
Defaults
system
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.2(2)XB
|
This command was introduced.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(8)T
|
This command was integrated into this release. The following were not supported: the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850.
|
12.2(11)T
|
This command was implemented on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
|
Usage Guidelines
There are two ways to configure reliable provisional responses:
•Dial-peer mode. You can configure reliable provisional responses for the specific dial peer only by using the voice-class sip rel1xx command.
•SIP mode. You can configure reliable provisional responses globally by using the rel1xx command.
The use of resource reservation with SIP requires that the reliable provisional feature for SIP be enabled either at the VoIP dial-peer level or globally on the router.
This command applies to the dial peer under which it is used or points to the global configuration for reliable provisional responses. If the command is used with the supported keyword, the SIP gateway uses the Supported header in outgoing SIP INVITE requests. If it is used with the require keyword, the gateway uses the Required header.
This command, in dial-peer configuration mode, takes precedence over the rel1xx command in global configuration mode with one exception: If this command is used with the system keyword, the gateway uses what was configured under the rel1xx command in global configuration mode.
Examples
The following example shows how to use this command on either an originating or a terminating SIP gateway:
•On an originating gateway, all outgoing SIP INVITE requests matching this dial peer contain the Supported header where value is 100rel.
•On a terminating gateway, all received SIP INVITE requests matching this dial peer support reliable provisional responses.
Router(config)# dial-peer voice 102 voip
Router(config-dial-peer)# voice-class sip rel1xx supported 100rel
Related Commands
Command
|
Description
|
rel1xx
|
Provides provisional responses for calls on all VoIP calls.
|
voice-class sip url
To configure URLs to either the Session Initiation Protocol (SIP) or telephone (TEL) format for your dial-peer SIP calls, use the voice-class sip url command in dial-peer configuration mode. To reset to the default, use the no form of this command.
voice-class sip url {sip | tel | system}
no voice-class sip url
Syntax Description
sip
|
Generates URLs in the SIP format for calls on a dial-peer basis.
|
tel
|
Generates URLs in the TEL format for calls on a dial-peer basis.
|
system
|
Uses the system value. This is the default.
|
Defaults
system
Command Modes
Dial-peer configuration
Command History
Release
|
Modification
|
12.2(2)XB
|
This command was introduced.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(8)T
|
This command was integrated into this release. The following were not supported: the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850.
|
12.2(11)T
|
This command was implemented on the Cisco AS5300, Cisco AS5350, and Cisco AS5400.
|
Usage Guidelines
This command affects only user-agent clients (UACs), because it causes the use of a TEL or SIP URL in the request line of outgoing SIP INVITE requests. SIP URLs indicate the originator, recipient, and destination of the SIP request; TEL URLs indicate voice-call connections.
The voice-class sip url command, in dial-peer configuration mode, takes precedence over the url command in SIP global-configuration mode. However, if the voice-class sip url command is used with the system keyword, the gateway uses what was globally configured under the url command.
Examples
The following example shows how to set up the voice-class sip url command to generate URLs in the TEL format:
Router(config)# dial-peer voice 102 voip
Router(config-dial-peer)# voice-class sip url tel
Related Commands
Command
|
Description
|
sip url
|
Generates URLs in the SIP or TEL format in VoIP configuration mode.
|
voice-encap
This command was added in Cisco IOS Release 11.3(1)MA on Cisco MC3810. This command is not supported in Cisco IOS Release 12.2.
voice-group
This command was added in Cisco IOS Release 11.3(1)MA for Cisco MC3810. This command is not supported in Cisco IOS Release 12.2.
voicemail (cm-fallback)
To configure the telephone number that is speed-dialed when the messages button on a Cisco IP phone is pressed, use the voicemail command in call-manager-fallback configuration mode. To disable the messages button, use the no form of this command.
voicemail phone-number
no voicemail
Syntax Description
phone-number
|
Phone number that is configured as a speed-dial number for retrieving messages.
|
Defaults
No phone number is configured, and the messages button is ineffective.
Command Modes
Call-manager-fallback configuration
Command History
Release
|
Modification
|
12.1(5)YD
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420.
|
12.2(2)XT
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
12.2(8)T
|
This command was implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3.
|
12.2(8)T1
|
This command was implemented on the Cisco 2600-XM and Cisco 2691.
|
12.2(11)T
|
This command was implemented on the Cisco 1760.
|
Usage Guidelines
This command configures the telephone number that is speed-dialed when the message button on a Cisco IP phone is pressed. The same voicemail telephone number is configured for all Cisco IP phones connected to the router.
Examples
The following example sets the phone number 4085551000 as the speed-dial number that is dialed to retrieve messages when the messages button is pressed:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# voicemail 914085551000
The number 914085551000 is called when the Cisco IP phone messages button is pressed to retrieve messages.
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables SRS Telephony feature support and enters call-manager-fallback configuration mode.
|
voicemail (telephony-service)
To configure the telephone number that is speed-dialed when the messages button on a Cisco IP phone is pressed, use the voicemail command in telephony-service configuration mode. To disable the messages button, use the no form of this command.
voicemail phone-number
no voicemail
Syntax Description
phone-number
|
Phone number that is configured as a speed-dial number for retrieving messages.
|
Defaults
No phone number is configured, and the messages button is ineffective.
Command Modes
Telephony-service configuration
Command History
Release
|
Modification
|
12.1(5)YD
|
This command was introduced on the Cisco 2600, Cisco 3600, and Cisco IAD2420.
|
12.2(2)XT
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
12.2(8)T
|
This command was implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
This command was implemented on the Cisco 2600-XM and Cisco 2691.
|
12.2(11)T
|
This command was implemented on the Cisco 1760.
|
Usage Guidelines
This command configures the telephone number that is speed-dialed when the messages button on a Cisco IP phone is pressed. The same telephone number is configured for voice mail for all Cisco IP phones connected to the router.
Examples
The following example sets the phone number 914085551000 as the speed-dial number that is dialed to retrieve messages when the messages button is pressed:
Router(config)# telephony-service
Router(config-telephony-service)# voicemail 914085551000
The number 914085551000 is called when the Cisco IP phone messages button is pressed to retrieve messages.
Related Commands
Command
|
Description
|
telephony-service
|
Enables Cisco IOS Telephony Service and enters telephony-service configuration mode.
|
vm-device-id (ephone)
|
Defines the voice-mail ID string.
|
voice-port
To enter voice-port configuration mode, use the voice-port command in global configuration mode.
Cisco 1750 and Cisco 1751
voice-port slot-number/port
Cisco 2600, Cisco 3600 Series and Cisco 7200 Series
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-no}
Cisco 2600 and Cisco 3600 Series with a High-Density Analog Network Module (NM-HDA)
voice-port {slot-number/subunit-number/port}
Cisco AS5300
voice-port controller-number:D
Cisco AS5800
voice-port {shelf/slot/port:D | shelf/slot/parent:port:D}
Cisco MC3810
voice-port slot/port
Syntax Description
Cisco 1750 and Cisco 1751
slot-number
|
Number of the slot in the router in which the voice interface card (VIC) is installed. Range is from 0 to 2, depending on the slot in which it is installed.
|
port
|
Voice port number. Range is from 0 to 1.
|
Cisco 2600, Cisco 3600 Series and Cisco 7200 Series
slot-number
|
Number of the slot in the router in which the VIC is installed. Range is from 0 to 3, depending on the slot in which it is installed.
|
subunit-number
|
Subunit on the VIC in which the voice port is located. Range is from 0 to 1.
|
port
|
Voice port number. Range is from 0 to 1.
|
slot
|
Router location in which the voice port adapter is installed. Range is from from 0 to 3.
|
port:
|
VIC location. Range is from 0 to 3.
|
ds0-group-no
|
Defined DS0 group number. Each such number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card.
|
Cisco AS5300:
controller-number
|
T1 or E1 controller.
|
:D
|
D channel associated with ISDN PRI.
|
Cisco AS5800:
shelf
|
T1 or E1 controller on the T1 card, or the T1 controller on the T3 card. Range is from 0 to 9999.
|
slot
|
T1 or E1 controller on the T1 card, or the T1 controller on the T3 card. Range is from 0 to 11.
|
port
|
Voice port number.
•T1 or E1 controller on the T1 card range is from 0 to 11.
•T1 controller on the T3 card range is from 1 to 28.
|
:port
|
Value for the parent argument. Valid entry is 0.
|
:D
|
D channel associated with ISDN PRI.
|
Cisco MC3810
slot
|
Slot in the router in which the VIC is installed. The only valid entry is 1.
|
port
|
Voice port number. Valid values are as follows:
•T1—ANSI T1.403 (1989), Bellcore TR-54016
•E1— ITU G.703
•Analog Voice—Up to six ports (FXS, FXO, E & M)
•Digital Voice— Single T1/E1 with cross-connect drop and insert, CAS and CCS signaling, PRI QSIG
•Ethernet—Single 10BASE T
•Serial—Two five-in-one synchronous serial (ANSI EIA/TA-530, EIA/TA-232, EIA/TA-449; ITU V.35, X.21, Bisync, Polled Async)
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(3)T
|
This command was implemented on the Cisco 2600 series.
|
12.0(3)T
|
This command was implemented on the Cisco AS5300.
|
12.0(7)T
|
This command was implemented on the Cisco AS5800, Cisco 7200 series, and Cisco 1750. Arguments were added for the Cisco 2600 series and Cisco 3600 series.
|
12.2(8)T
|
This command was implemented on the Cisco 1751 and Cisco 1760. The command was modified to accommodate the additional ports of the NM-HDA on the Cisco 2600 series, Cisco 3640, and Cisco 3660.
|
12.2(2)XN
|
Support for enhanced MGCP voice gateway interoperability was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.
|
12.2(11)T
|
This command was integrated into Cisco CallManager Version 3.2 and implemented on the Cisco IAD2420 series.
|
12.2(13)T
|
This command was integrated into this release. The following was not supported: the extended echo canceller (EC) feature on the Cisco AS5300 and Cisco AS5800.
|
Usage Guidelines
Use the voice-port global configuration command to switch to voice-port configuration mode from global configuration mode. Use the exit command to exit voice-port configuration mode and return to global configuration mode.
Note This command does not support the extended echo canceller (EC) feature on the Cisco AS5300 or the Cisco AS5800.
Examples
The following example accesses voice-port configuration mode for port 0, located on subunit 0 on a VIC installed in slot 1 of a Cisco 3600 series router:
The following example accesses voice-port configuration mode for digital voice port 24 on a Cisco MC3810 that has a digital voice module (DVM) installed:
The following example accesses voice-port configuration mode for a Cisco AS5300:
The following example accesses voice-port configuration mode for a Cisco AS5800 (T1 card):
The following example accesses voice-port configuration mode for a Cisco AS5800 (T3 card):
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial-peer configuration mode and specifies the method of voice encapsulation.
|
voice-port (MGCP profile)
The voice-port (MGCP profile) command is replaced by the port (MGCP profile) command in Cisco IOS Release 12.2(8)T. See the port (MGCP profile) command for more information.
voice-port busyout
To place all voice ports associated with a serial or ATM interface into a busyout state, use the voice-port busyout command in interface configuration mode. To remove the busyout state on the voice ports associated with this interface, use the no form of this command.
voice-port busyout
no voice-port busyout
Syntax Description
This command has no arguments or keywords.
Defaults
The voice ports on the interface are not in busyout state.
Command Modes
Interface configuration
Command History
Release
|
Modification
|
12.0(3)T
|
This command was introduced on the Cisco MC3810.
|
Usage Guidelines
This command busies out all voice ports associated with the interface, except any voice ports configured to busy out under specific conditions using the busyout monitor and busyout seize commands.
Examples
The following example places the voice ports associated with serial interface 1 into busyout state:
interface serial 1
voice-port busyout
The following example places the voice ports associated with ATM interface 0 into busyout state:
Related Commands
Command
|
Description
|
busyout forced
|
Forces a voice port on the Cisco MC3810 into the busyout state.
|
busyout monitor
|
Places a voice port on the Cisco MC3810 into the busyout monitor state.
|
busyout seize
|
Changes the busyout action for an FXO or FXS voice port.
|
show voice busyout
|
Displays information about the voice busyout state on the Cisco MC3810.
|
voip-incoming translation-profile
To specify a translation profile for all incoming VoIP calls, use the voip-incoming translation-profile command in global configuration mode. To delete the profile, use the no form of this command.
voip-incoming translation-profile name
no voip-incoming translation-profile name
Syntax Description
name
|
Name of the translation profile.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(11)T
|
This command was introduced.
|
Usage Guidelines
Use the voip-incoming translation-profile command to globally assign a translation profile for all incoming VoIP calls. The translation profile was previously defined using the voice translation-profile command. The voip-incoming translation-profile command does not require additional steps to complete its definition.
If an H.323 call comes in and the call is associated with a source IP group that is defined with a translation profile, the source IP group translation profile overrides the global translation profile.
Examples
The following example assigns the translation profile named "global-definition" to all incoming VoIP calls:
Router(config)# voip-incoming translation-profile global-definition
Related Commands
Command
|
Description
|
show voice translation-profile
|
Displays the configurations for all voice translation profiles.
|
test voice translation-rule
|
Tests the voice translation rule definition.
|
voice translation-profile
|
Initiates a translation profile definition.
|
voip-incoming translation-rule
To set the incoming translation rule for calls that originate from H.323-compatible clients, use the voip-incoming translation-rule command in global configuration mode. To disable the incoming translation rule, use the no form of this command.
voip-incoming translation-rule tag {calling-number | called-number}
no voip-incoming translation-rule tag {calling-number | called-number}
Syntax Description
tag
|
Tag number by which the rule set is referenced. This is an arbitrarily chosen number. Range is from 1 to 2147483647. There is no default value.
|
calling-number
|
Automatic number identification (ANI) number or the number of the calling party.
|
called-number
|
Dial Number Information Service (DNIS) number or the number of the called party.
|
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XR1
|
This command was introduced for VoIP on the Cisco AS5300.
|
12.0(7)XK
|
This command was implemented for VoIP on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.1(1)T
|
This command was implemented for VoIP on the Cisco 1750, Cisco AS5300, Cisco 7200, and Cisco 7500.
|
12.1(2)T
|
This command was implemented for VoIP on the Cisco MC3810.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(11)T
|
This command was integrated into this release.
|
Usage Guidelines
With this command, all IP-based calls are captured and handled, depending on either the calling number or the called number to the specified tag name.
Examples
The following example identifies the rule set for calls that originate from H.323-compatible clients:
Router(config)# voip-incoming translation-rule 5 called-number
Related Commands
Command
|
Description
|
numbering-type
|
Matches one number type for a dial-peer call leg.
|
rule
|
Applies a translation rule to a calling party number or a called party number for both incoming and outgoing calls.
|
show translation-rule
|
Displays the contents of all the rules that have been configured for a specific translation name.
|
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.
|
volume
To set the receiver volume level for a POTS port on a router, use the volume command in dial-peer voice configuration mode. To reset to the default, use the no form of this command.
volume number
no volume number
Syntax Description
number
|
Decibels (dB) of gain. Range is as follows:
•1: -11.99 dB
•2: -9.7dB
•3: -7.7dB
•4: -5.7dB
•5: -3.7dB
Default is 3 (-7.7 dB gain).
|
Defaults
3 (-7.7 dB gain)
Command Modes
Dial-peer voice configuration
Command History
Release
|
Modification
|
12.2(8)T
|
This command was introduced on the Cisco 803, Cisco 804, and Cisco 813.
|
Usage Guidelines
Set the volume command for each POTS port separately. Setting the volume level affects only the port for which it has been set.
Note Only the receiver volume is set with this command.
Use the show pots volume command to check the volume status and level.
Examples
The following example shows a volume level of 4 for POTS port 1 and a volume level of 2 for POTS port 2.
destination-pattern 5551111
destination-pattern 5552222
Related Commands
Command
|
Description
|
show pots volume
|
Shows the receiver volume configured for each POTS port on a router.
|
web admin customer
To define a username and password for a Cisco IOS Telephony System (ITS) customer administrator, use the web admin customer command in telephony-service configuration mode. To disable a customer administrator login, use the no form of this command.
web admin customer name username {password string | secret {0 | 5} string}
no web admin customer
Syntax Description
name username
|
Username for the customer administrator. Default is Customer.
|
password string
|
Password for the customer administrator. Default is no password.
|
secret {0 | 5} string
|
Secret password and whether or not it is encrypted. Keywords are as follows:
•0—Password that follows is not encrypted.
•5—Password that follows is encrypted.
|
Defaults
A customer administrator named Customer with no password is defined.
Command Modes
Telephony-service
Command History
Release
|
Modification
|
12.2(11)YT
|
This command was introduced.
|
12.2(15)T
|
This command was integrated into this release.
|
Usage Guidelines
Use this command with Cisco IOS Telephony Service (ITS) V2.1 or a later version.
Examples
The following example defines a customer administrator named user22 whose password is pw567890:
Router(config)# telephony-service
Router(config-telephony-service)# web admin customer name user22 password pw567890
Related Commands
Command
|
Description
|
telephony-service
|
Enables Cisco ITS and enters telephony-service configuration mode.
|
web customize load
|
Loads and parses an eXtensible Markup Language (XML) file in router Flash memory to customize a graphical user interface (GUI) for a customer administrator using Cisco ITS.
|
web admin system
To define a username and password for a Cisco IOS Telephony Service (ITS) system administrator, use the web admin system command in telephony-service configuration mode. To disable a customer administrator login, use the no form of this command.
web admin system name username {password string | secret {0 | 5} string}
no web admin system
Syntax Description
name username
|
Login name for the system administrator. Default is Admin.
|
password string
|
Character string for login authentication, stored in the running configuration as plain text. Default is no password.
|
secret {0 | 5} string
|
Character string for login authentication, stored in the running configuration as encrypted using MD5, and whether or not it is encripted. Keywords are as follows:
•0—Password that follows is not encrypted.
•5—Password that follows is encrypted.
|
Defaults
A system administrator named Admin with no password is defined.
Command Modes
Telephony-service
Command History
Release
|
Modification
|
12.2(11)YT
|
This command was introduced.
|
12.2(15)T
|
This command was integrated into this release.
|
Usage Guidelines
Use this command with Cisco ITS V2.1 or a later version.
You can encrypt the system administrator password with MD5 by using the secret 0 keyword pair before entering a plain-text password string. An encrypted version of the string is saved in the running configuration, as shown in the following example. Note that the digit 5 appears in this line to indicate that the password that follows is shown in its encrypted version.
web admin system name jsmith secret 5 $1$TCyK$OU/NSQ/VtAU2ibHdi8Uau
Examples
The following example establishes a system administrator named user1 whose password is pw234567:
Router(config)# telephony-service
Router(config-telephony-service)# web admin system name user1 password pw234567
Related Commands
Command
|
Description
|
telephony-service
|
Enables Cisco ITS and enters telephony-service configuration mode.
|
web customize load
To load and parse an eXtensible Markup Language (XML) file in router Flash memory to customize a graphical user interface (GUI) for a customer administrator using Cisco IOS Telephony Service (ITS), use the web customize load command in telephony-service configuration mode. To disable the customized GUI and fall back to the system administrator GUI, use the no form of this command.
web customize load filename
no web customize load
Syntax Description
filename
|
XML file in flash memory that is to be loaded and parsed. This file defines the customer administrator GUI.
|
Defaults
The standard system administrator GUI is used.
Command Modes
Telephony-service configuration
Command History
Release
|
Modification
|
12.2(11)YT
|
This command was introduced.
|
12.2(15)T
|
This command was integrated into this release.
|
Usage Guidelines
Use this command with Cisco ITS V2.1 or a later version.
Examples
The following example specifies a file named cust_admin_gui.xml as the file that defines the GUI for ITS customer administrators:
Router(config)# telephony-service
Router(config-telephony-service)# web customize load cust_admin_gui.xml
Related Commands
Command
|
Description
|
telephony-service
|
Enables Cisco ITS and enters telephony-service configuration mode.
|