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Cisco IOS Voice Command Reference
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About Cisco IOS Software Documentation
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Using Cisco IOS Software
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Commands: A
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Commands: B
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Commands:cac master through call application stats
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Commands: call application voice through call-denial
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Commands: call fallback through called-number (dial-peer)
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Commands: caller-id (dial peer) through ccm-manager switchover-to-backup
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Commands: ccs connect (controller) through clear vsp statistics
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Commands: clid through cptone
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Commands: default (MGCP profile) through direct-inward-dial
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Commands: disable-early-media 180 through dualtone
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Commands: E
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Commands: F
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Commands: G
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Commands: H
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Commands: icpif through irq global-request
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Commands: isdn bind-l3 through ixi transport http
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Commands: K
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Commands: L
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Commands: map q850-cause through mgcp package-capability
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Commands: mgcp persistent through mmoip aaa send-id secondary
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Commands: mode (ATM/T1/E1 controller) through mwi-server
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Commands: N
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Commands: O
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Commands: package through pattern
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Commands: permit hostname through proxy h323
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Commands: Q
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Commands: R
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Commands: sccp through service-type call-check
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Commands: session through sgcp tse payload
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Commands: show aal2 profile through show call filter match-list
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Commands: show call history fax through show debug condition
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Commands: show dial-peer video through show gatekeeper zone status
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Commands: show gateway through show modem relay statistics
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Commands: show mrcp client session active through show sgcp statistics
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Commands: show sip service through show trunk hdlc
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Commands: show vdev through show voice statistics memory-usage
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Commands: show voice translation-profile through shutdown (voice-port)
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Commands: signal through srv version
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Commands: ss7 mtp2-variant through switchover method
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Commands: target carrier-id through threshold noise
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Commands: timeout (auto-config application) through timing clear-wait
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Commands:timing delay-duration through type (voice)
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Commands: U
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Commands: vad (dial peer) through voice class dualtone-detect-params
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Commands: voice class h323 through vxml tree memory
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Commands: W
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Commands: Z
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Table Of Contents
voice-class permanent (dial peer)
voice-class permanent (voice-port)
voice-class sip outbound-proxy
voice-class sip resource priority mode (dial peer)
voice-class sip resource priority namespace (dial peer)
voice-class sip transport switch
voice class uri sip preference
voice statistics accounting method
voice statistics display-format separator
voice statistics max-storage-duration
voip-incoming translation-profile
voip-incoming translation-rule
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
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 demonstrates how a voice class is created and applied to an individual dial peer. Voice class 4 contains a command to disable the capability to detect Cisco CallManager systems in the network (this command is used by Cisco CallManager Express 3.1 and later versions). The example then uses the voice-class h323 command to apply voice class 4 to dial peer 36.
Router(config)# voice class h323 4Router(config-class)# no telephony-service ccm-compatibleRouter(config-class)# exitRouter(config)# dial-peer voice 36 voipRouter(config-dial-peer)# destination-pattern 555....Router(config-dial-peer)# session target ipv4:10.5.6.7Router(config-dial-peer)# voice-class h323 4Related Commands
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
Command Default
The dial peer does not use an H.323 voice class.
Command Modes
Dial peer configuration
Command History
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 demonstrates how a voice class is created and applied to an individual dial peer. Voice class 4 contains a command to disable the capability to detect Cisco CallManager systems in the network (this command is used by Cisco CallManager Express 3.1 and later versions). The example then uses the voice-class h323 command to apply voice class 4 to dial peer 36.
Router(config)# voice class h323 4Router(config-class)# no telephony-service ccm-compatibleRouter(config-class)# exitRouter(config)# dial-peer voice 36 voipRouter(config-dial-peer)# destination-pattern 555....Router(config-dial-peer)# session target ipv4:10.5.6.7Router(config-dial-peer)# voice-class h323 4Related Commands
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
Command Default
No voice class is configured.
Command Modes
Global configuration
Command History
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:
voice class permanent 10signal keepalive 3exitRelated Commands
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 number 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.
Command Default
Network dial peers have no voice class assigned.
Command Modes
Dial peer configuration
Command History
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:
dial-peer voice 100 vofrvoice-class permanent 10Related Commands
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 number 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.
Command Default
Voice ports have no voice class assigned.
Command Modes
Voice-port configuration
Command History
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 1/1/0:
voice-port 1/1/0voice-class permanent 10Related Commands
voice-class sip e911
To enable SIP E911 system services on a dial peer, use the voice-class sip e911 command in VoIP dialpeer configuration mode. To disable SIP E911 services, use the no form of this command.
voice-class sip e911
no voice-class sip e911
Syntax Description
This command has no arguments or keywords.
Command Default
The dial peer uses the global setting.
Command Modes
VoIP dialpeer configuration mode.
Command History
Usage Guidelines
The no form of this command sets the dial peer configuration to disable, which indicates that E911 will not be used for this peer. Because the no version of the command causes non default behavior, it can been seen in the show running-config output. See also the voice service voip sip e911 and debug csm neat commands.
Examples
The following examples enable and disable E911 services on a VoIP dial peer:
Router(config)# dial-peer voice 2Router(config-dial-peer)# voice-class sip e911*Jun 06 00:47:20.611: setting peer 2 to enableRouter(config-dial-peer)# no voice-class sip e911*Jun 06 00:49:58.931: setting peer 2 to disableRelated Commands
voice-class sip localhost
To define a local hostname used to generate gateway URLs, use the voice-class sip localhost command in global configuration mode. To disable a local hostname, use the no form of this command.
voice-class sip localhost [dns]:local-host-name-string
no voice-class sip localhost [dns]:local-host-name-string
Syntax Description
dns
(Optional) Domain name server.
:
Command delimiter.
local-host-name-string
Name used to generate gateway URLs.
Command Default
No local hostnames used for locally generated gateway URLs are sent.
Command Modes
Global configuration
Command History
Usage Guidelines
The no voice-class sip localhost command causes the calls using that dial peer to default to the base behavior (use IP addresses in the host portion of the From, Call-ID, and Remote-Party-ID headers). This allows the administrator to turn off this feature on selected dial peers.
Examples
The following example defines a local hostname used to generate a gateway URL:
Router(config)# voice-class sip localhost dns:gateway2.customer1.comRelated Commands
dial-peer voice
Defines a particular dial peer, to specify the method of voice encapsulation, and enters dial peer configuration mode.
voice-class sip outbound-proxy
To configure an outbound proxy, use the voice-class outbound-proxy command in dial-peer configuration mode. To reset to, use the no form of this command.
voice-class sip outbound-proxy {ipv4: ip address | dns: host:domain}
no voice-class sip outbound-proxy
Syntax Description
Command Default
This command is disabled.
Command Modes
Voice-service sip configuration (conf-serv-sip)
Dial-peer configuration (config-dial-peer)Command History
Usage Guidelines
The voice-class sip outbound-proxy command, in dial-peer configuration mode, takes precedence over the command in SIP global-configuration mode.
Examples
The following example shows how to set up the voice-class sip outbound-proxy command globally to generate:
Router> enableRouter# configure terminalRouter(config)# voice service voipRouter(conf-voi-serv)# sipRouter(conf-serv-sip)# no outbound-proxy ipv4:10.1.1.1The following example shows how to set up the voice-class sip outbound-proxy command globally to generate:
Router> enableRouter# configure terminalRouter(config)# voice service voipRouter(conf-voi-serv)# sipRouter(conf-serv-sip)# no outbound-proxy dns:sipproxy:cisco.comThe following example shows how to set up the voice-class sip outbound-proxy command on a dial peer to generate:
Router> enableRouter# configure terminalRouter(conf)# dial-peer voice 111 voipRouter(config-dial-peer)# voice-class sip outbound-proxy ipv4:10.1.1.1The following example shows how to set up the voice-class sip outbound-proxy command on a dial peer to generate:
Router> enableRouter# configure terminalRouter(conf)# dial-peer voice 111 voipRouter(config-dial-peer)# voice-class sip outbound-proxy dns:sipproxy:cisco.comRelated Commands
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
Command Default
system
Command Modes
Dial peer configuration
Command History
Usage Guidelines
There are two ways to configure reliable provisional responses:
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Dial-peer mode. You can configure reliable provisional responses for the specific dial peer only by using the voice-class sip rel1xx command.
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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:
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Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip rel1xx supported 100relRelated Commands
voice-class sip resource priority mode (dial peer)
To push the user access server (UAS) to operate in a loose or strict mode, use the voice-class sip resource priority mode command in dial peer voice configuration mode. To disable the voice-class sip resource priority mode, use the no form of this command.
voice-class sip resource priority mode [loose | strict]
no voice-class sip resource priority mode [loose | strict]
Syntax Description
Command Default
The default value is loose mode.
Command Modes
Dial peer voice configuration
Command History
Usage Guidelines
When the no version of this command is executed, the call operates in the loose mode.
Examples
The following example shows how to set up the voice-class sip resource priority mode command in loose mode:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip resource priority mode looseThe following example shows how to set up the voice-class sip resource priority mode command in strict mode:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip resource priority mode strictRelated Commands
voice-class sip resource priority namespace
Priorities mandatory call prioritization handling for initial original INVITE message requests.
voice-class sip resource priority namespace (dial peer)
To prioritize mandatory call prioritization handling for initial original INVITE message requests, use the voice-class sip resource priority namespace command in dial peer voice configuration mode. To disable the voice-class sip resource priority namespace command, use the no form of this command.
voice-class sip resource priority namespace [drsn | dsn | q735]
no voice-class sip resource priority namespace [drsn | dsn | q735]
Syntax Description
Command Default
When the no version of this command is executed using namespace, the Cisco IOS gateway transparently passes the multilevel precedence and preemption (MLPP) values that were received on the PSTN side.
Command Modes
Dial peer voice configuration
Command History
Usage Guidelines
When the no version of this command is executed using the namespace, the Cisco IOS gateway transparently passes the multilevel precedence and preemption (MLPP) values that were received on the PSTN side.
Examples
The following example shows how to set up the voice-class sip resource priority namespace command in the U. S. DSN format name space:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip resource priority namespace dsnThe following example shows how to set up the voice-class sip resource priority namespace command in the U. S. DRSN format name space:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip resource priority namespace drsnThe following example shows how to set up the voice-class sip resource priority namespace command in the Public SS7 Network format name space:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip resource priority namespace q735Related Commands
voice-class sip resource priority mode
Pushes the UAS to operate in a loose or strict mode.
voice-class sip transport switch
To enable switching between UDP and TCP transport mechanisms for large Session Initiation Protocol (SIP) messages for a specific dial peer, use the voice-class sip transport switch command in dial peer configuration mode. To disable switching between UDP and TCP transport mechanisms for large SIP messages for a specific dial peer, use the no form of this command.
voice-class sip transport switch udp tcp
no voice-class sip transport switch udp tcp
Syntax Description
udp
Enables switching transport from UDP on the basis of the size of the SIP request being greater than the MTU size.
tcp
Enables switching transport to TCP.
Command Default
Disabled.
Command Modes
Dial peer configuration
Command History
Usage Guidelines
The voice-class sip transport switch command takes precedence over the global transport switch command.
Examples
The following example shows how to set up the voice-class sip transport switch command:
Router(config)# dial-peer voice 102 voipRouter(config-dial-peer)# voice-class sip transport switch udp tcpRelated Commands
voice-class sip url
To configure URLs to either the Session Initiation Protocol (SIP), SIP security (SIPS), 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 value (system), use the no form of this command.
voice-class sip url {sip | sips | system | tel}
no voice-class sip url
Syntax Description
Command Default
system
Command Modes
Dial peer configuration
Command History
Usage Guidelines
This command affects only user-agent clients (UACs), because it causes the use of a SIP, SIPS, or TEL 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 SIP format:
dial-peer voice 102 voipvoice-class sip url sipThe following example shows how to set up the voice-class sip url command to generate URLs in the SIPS format:
dial-peer voice 102 voipvoice-class sip url sipsThe following example shows how to set up the voice-class sip url command to generate URLs in the TEL format:
dial-peer voice 102 voipvoice-class sip url telRelated Commands
sip url
Generates URLs in the SIP, SIPS, or TEL format.
url
Configures URLs to either session initiation protocol (SIP), SIP secure (SIPS), or telephone (TEL) format.
voice class tone-signal
To enter voice-class configuration mode and create a tone-signal voice class, use the voice class tone-signal command in global configuration mode. To delete a tone-signal voice class, use the no form of this command.
voice class tone-signal tag
no voice class tone-signal tag
Syntax Description
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
Use the voice class tone-signal command to define wakeup, frequency selection, and guard tones to be played out before and during the voice packets for a specific voice port. Use the inject guard-tone, inject pause, and inject tone commands to define the tone signaling in this class. You can configure up to ten tones in a tone-signal voice class.
To avoid voice loss at the receiving end of an LMR system, the maximum of the sum of the durations of the injected tones and pauses in the voice class should not exceed 1500 milliseconds. You must also use the timing delay-voice tdm command to configure a delay for the voice packet equal to the sum of the durations of all the injected tones and pauses.
Note that the hyphenation in this command differs from the hyphenation used in a similar command, voice-class tone-signal, which is used in voice-port configuration mode.
Examples
The following example shows how to create a tone-signal voice class starting from global configuration mode:
voice class tone-signal mytonesinject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50Related Commands
voice-class tone-signal
To assign a previously configured tone-signal voice class to a voice port, use the voice-class tone-signal command in voice-port configuration mode. To delete a tone-signal voice class, use the no form of this command.
voice-class tone-signal tag
no voice-class tone-signal tag
Syntax Description
tag
Unique label assigned to the voice class. The tag label maps to the tag label created using the voice class tone-signal global configuration command. Can be up to 32 alphanumeric characters.
Command Default
Voice ports have no tone-signal voice class assigned.
Command Modes
Voice-port configuration
Command History
12.3(4)XD
This command was introduced.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
Usage Guidelines
The voice-class tone-signal command is available on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Note that the hyphenation in this command differs from the hyphenation used in a similar command, voice class tone-signal, which is used in global configuration mode.
Examples
The following example assigns a previously configured voice class to voice port 1/1/0:
voice-port 1/0/0voice-class tone-signal mytonesRelated Commands
voice class tone-signal
Enters voice-class configuration mode and assigns an identification tag number for a tone-signal voice class.
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.
Command Default
The two-beep confirmation tone is heard on PLAR and PLAR OPX connections.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
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 1/0/0:
voice-port 1/0/0connection plar-opxvoice confirmation-toneRelated Commands
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.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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.
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•
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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).
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•
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Examples
The following example shows how the voice dnis-map command is used to create a DNIS map:
voice dnis-map dmap1The 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.txtFollowing is an example of the contents of a text file comprising a DNIS map:
!Example dnis-map with 8 entries.!dnis 5550112 url tftp://global/ticket/vapptest1.vxmldnis 5550111 url tftp://global/ticket/vapptest2.vxmldnis 5550134 url tftp://global/ticket/vapptest3.vxmldnis 5550178dnis 5550100dnis 5550101dnis 5550102dnis 5550103Related Commands
voice class uri
To create or modify a voice class for matching dial peers to a Session Initiation Protocol (SIP) or telephone (TEL) uniform resource identifier (URI), use the voice class uri command in global configuration mode. To remove the voice class, use the no form of this command.
voice class uri tag {sip | tel}
no voice class uri tag
Syntax Description
tag
Label that uniquely identifies the voice class. Can be up to 32 alphanumeric characters.
sip
Voice class for SIP URIs.
tel
Voice class for TEL URIs.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
Usage Guidelines
•
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Examples
The following example defines a voice class for SIP URIs:
voice class uri r100 sipuser-id abc123host server1phone context 408The following example defines a voice class for TEL URIs:
voice class uri r101 telphone number ^408phone context 408Related Commands
voice class uri sip preference
To set the preference for selecting a voice class for Session Initiation Protocol (SIP) uniform resource identifiers (URIs), use the voice class uri sip preference command in global configuration mode. To reset to the default, use the no form of this command.
voice class uri sip preference {user-id | host}
no voice class uri sip preference
Syntax Description
Command Default
Host field
Command Modes
Global configuration
Command History
Usage Guidelines
•
•
Examples
The following example defines the preference as the user-id for a SIP voice class:
voice class uri sip preference user-idRelated Commands
voice dnis-map load
To reload a DNIS map that has been modified, use the voice dnis-map load command in privileged EXEC mode. This command does not have a no form.
voice dnis-map load map-name
Syntax Description
Command Default
No default behavior or values
Command Modes
Privileged EXEC
Command History
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 reloads a DNIS map named "mapfile1":
Router# voice dnis-map load mapfile1Related Commands
voice dsp crash-dump
To enable the crash dump feature and to specify the destination file and the file limit, enter the
voice dsp crash-dump command in global configuration mode. To disable the feature, use the no form of the command.voice dsp crash-dump [destination url | file-limit limit-number]
no voice dsp crash-dump
Syntax Description
Command Default
Crash dump capability is turned off.
Command Modes
Global configuration
Command History
Usage Guidelines
To configure the router to write a crash dump file, the destination url in the voice dsp crash-dump command must be set to a valid file system, and the crash dump file limit must be set to a non-zero value. The default is that the crash dump capability is turned off, as the url field is empty, and the file limit is zero.
As each crash-dump file is created, the name of the file has a number appended to the end. This number is incremented from 1 to up to the file limit for each subsequent crash dump file written. If the router reloads, the number is reset back to 1, and so file number 1 is written again. After the file number reaches the maximum file limit, no more files are written.
The file count can be manually reset by setting the file limit to zero and then setting it to a non-zero limit. This has the effect of restarting the count of files written, causing the files 1 to the file limit of 99 to be able to be written again, thus overwriting the original files.
Setting the file-number argument to zero (the default) disables the collection of the dump from the DSP. In this case, the memory is not collected from the DSP, and the stack is not displayed on the console. If the keepalive mechanism detects a crashed DSP, the DSP is simply restarted.
Setting the file-number argument to a non-zero number but having a null destination url causes the dump to be collected and the stack to be displayed on the console, but no dump file is written.
If auto-recovery is turned off for the router, no DSP dump functions are enabled, no keepalive checks are done, and no dumps are collected or written.
Note
Note
Examples
The following example enables the crash dump feature and specifies the destination file in slot 0:
Router configure terminalEnter configuration commands, one per line. End with CNTL/Z.Router(config)# voice dsp crash-dump destination slot0:banjo-152-sRouter# end1w0d:%SYS-5-CONFIG_I:Configured from console by consoleCheck your configuration by entering the show voice dsp crash-dump command in privileged EXEC configuration mode:
Router# show voice dsp crash-dumpVoice DSP Crash-dump status:Destination file url is slot0:banjo-152-sFile limit is 20Last DSP dump file written wastftp://112.29.248.12/tester/26-152-t2Next DSP dump file written will be slot0:banjo-152-s1Related Commands
debug voice dsp crash-dump
Displays crash dump debug information.
show voice dsp crash-dump
Displays voice dsp crash dump information.
voice echo-canceller extended
To enable the extended G.168 echo canceller (EC) on the Cisco 1700 series, Cisco ICS7750, or Cisco AS5300, use the voice echo-canceller extended command in global configuration mode. To reset to the default, use the no form of this command.
Cisco 1700 series and Cisco ICS 7750
voice echo-canceller extended
no voice echo-canceller extended
Cisco AS5300
voice echo-canceller extended [codec small codec large codec]
no voice echo-canceller extended
Syntax Description
Command DefaultV
Proprietary Cisco G.165 EC is enabled.
Command Modes
Global configuration
Command History
12.2(13)T
This command was introduced.
12.3(3)
This command was modified to allow unrestricted codecs on the Cisco AS5300. The codec keyword was made optional.
Usage Guidelines
Cisco 1700 series and Cisco ICS7750
You do not have to shut down all the voice ports on the Cisco 1700 series or Cisco ICS7750 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, you should disable this command if it is not needed.
Cisco AS5300
This command is available only on the Cisco AS5300 with C542 or C549 digital signal processor module (DSPM) high-complexity firmware.
The voice echo-canceller extended command enables the extended EC on a Cisco AS5300 using C549 DSP firmware with one channel of voice per DSP and unrestricted codecs. Any codec is supported.
The voice echo-canceller extended codec command enables the extended EC on a Cisco AS5300 using C542 or C549 DSP firmware with two channels of voice per DSP and restricted codecs. Only specific codecs can be used with the extended EC.
If fax-relay is not selected as the large codec, the VoIP dial peer requires that you use the
fax rate disabled command in dial peer configuration mode.After choosing the codecs to be supported by the extended echo canceller, either remove all dial peers with different codecs not supported by your new configuration or modify the dial-peer codec selection by selecting a voice codec or fax-relay. When codecs are restricted, only one selection is allowed. You must have a VoIP dial peer configured with an extended EC-compatible codec to ensure voice quality on the connection.
This command is not accepted if there are active calls. If the EC is already in effect and a codec choice is changed, the system scans the dial peers. Any dial peers that do not conform to the new global command settings are changed, and the user is informed of the changes. Similarly, modem relay is incompatible with the extended EC and must be disabled globally for all dial peers.
Note
Examples
The following example sets the extended G.168 EC on the Cisco 1700 series or Cisco ICS7750:
Router(config)# voice echo-canceller extendedThe following example sets the extended G.168 EC on the Cisco AS5300 with restricted codecs:
Router(config)# voice echo-canceller extended codec small g711 large g726The following example shows an error message that displays when a restricted codec is not allowed:
Cannot configure now, dial-peer 8800 is configured with codec=g728, fax rate=disable, modem=passthrough system.If necessary set this command to 'no', re-configure dial-peer codec, fax rate and/or modem. Then re-enter this command.In the above example, dial peer 8800 is misconfigured with a codec type, g728, that was not selected for the large codec type using the voice echo-canceller extended command.
Related Commands
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
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 3Router(config-enum)# rule 1 5/(.*)/ /\1/e164.cisco.comRouter(config-enum)# rule 2 4/^9011\(.*\)/ /\1/e164.arpaIn 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
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 328, 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 HPI capture buffer, in bytes. Range is from 328 to 9000000. The default is 328.
destination url
(Optional) Destination URL for storing captured data.
Command Default
328 bytes (no buffer is used if it is not configured explicitly)
Command Modes
Global configuration
Command History
12.2(10)
This command was introduced.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
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
Examples
The following example changes the size (in bytes) of the HPI capture buffer and initializes the buffer-queueing program:
Router# configure terminalEnter configuration commands, one per line. End with CNTL/Z.Router(config)# voice hpi capture buffer 40000Router(config)# endRouter#03:23:31:caplog:caplog_cli_interface:hpi capture buffer size set to 40000 bytes03:23:31: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-38-11703:23:32:%SYS-5-CONFIG_I:Configured from console by consoleRouter#The following example sets the capture destination by entering a destination URL using FTP:
Router# configure terminalEnter configuration commands, one per line. End with CNTL/Z.Router(config)# voice hpi capture destination ftp://172.23.184.233/c:b-38-117aRouter(config)#04:05:10:caplog:caplog_cli_interface:hpi capture destination:ftp://172.23.184.233/c:b-38-117a04:05:10:caplog:caplog_logger_init:TRUE, Started task HPI Logger (PID 19)04:05:10:caplog:caplog_cache_init:Cache must be at least 324 bytes04:05:10:caplog:caplog_logger_proc:Terminating...Router(config)# endRouter#Related Commands
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 specified disconnect cause 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 specified disconnect cause code (the default condition), use the no form of this command. To restore the default dial-peer hunt setting, use the default form of this command.
voice hunt {disconnect-cause-code | all}
no voice hunt {disconnect-cause-code | all}
default voice hunt
Syntax Description
disconnect-cause-code
A code returned from the destination router to indicate why an attempted end-to-end call was unsuccessful. If the specified disconnect cause code is returned from the last destination endpoint, dial peer hunting is enabled or disabled. Table 242 in the "Usage Guidelines" section describes the possible values. You can enter the keyword, decimal value, or hexadecimal value.
all
Continue dial-peer hunting for all disconnect cause codes returned from the destination endpoint.
default
Restores the default dial-peer hunt setting, that is, the router stops dial-peer hunting if it receives the user-busy or no-answer disconnect cause code.
Command Default
The router stops dial-peer hunting if it receives the user-busy or no-answer disconnect cause code.
Command Modes
Global configuration
Command History
Usage Guidelines
This command is used with routers that act as originating or tandem nodes in a VoIP, VoFR, or Voice over ATM environment.
For an outgoing call from an originating VoIP gateway configured for rotary dial-peer hunting, more than one dial peer may match the same destination number. The matching dial peers may have different routes. After the voice call using the first dial peer gets disconnected, it will return a disconnect cause code. To have the router to pick up the next matching dial peer in the rotary group and set up a call, the router must be configure to continue hunting the various routes. Use this command to configure the router's hunting behavior when specified cause codes are received.
You can use this command to enable and disable dial-peer hunting when nonstandard disconnect cause codes are received. Nonstandard disconnect cause codes are those that are not defined in ITU-T Recommendation Q.931, but are used by service providers. When this command is used to disable dial-peer hunting for a specific disconnect cause code, it appears in the running configuration of the router.
The disconnect cause codes are described in Table 242. The decimal and hexadecimal value of the disconnect cause code follows the description of each possible keyword.
Examples
The following example configures the originating or tandem router to continue dial-peer hunting if it receives a user-busy disconnect cause code from a destination router:
voice hunt user-busyThe following example configures the originating or tandem router to continue dial-peer hunting if it receives an invalid-number disconnect cause code from a destination router:
voice hunt 28The following example configures the originating or tandem router to continue dial-peer hunting if it receives a facility-not-subscribed disconnect cause code from a destination router:
voice hunt 0x32Related Commands
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 iec syslog
To enable viewing of Internal Error Codes as they are encountered in real time, use the voice iec syslog command in global configuration mode. To disable IEC syslog messages, use the no form of this command.
voice iec syslog
no voice iec syslog
Syntax Description
This command has no arguments or keywords.
Command Default
IEC syslog messages are disabled.
Command Modes
Global configuration
Command History
Examples
The following example enables IEC syslog messages:
router(config)# voice iec syslogRelated Commands
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.
Command Default
Local calls bypass the DSP.
Command Modes
Global configuration
Command History
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 router to pass local calls through the DSP:
no voice local-bypassRelated Commands
input gain
Configures a specific input gain value.
output attenuation
Configures a specific output attenuation value.
voicemail (stcapp-fsd)
To designate an SCCP telephony control (STC) application feature speed-dial code to speed dial the voice-mail number, use the voicemail command in STC application feature speed-dial configuration mode. To return the code to its default, use the no form of this command.
voicemail keypad-character
no voicemail
Syntax Description
Command Default
The default voice-mail code is 0 (zero) for one-digit codes; 00 (two zeros) for two-digit codes.
Command Modes
STC application feature speed-dial configuration
Command History
12.4(2)T
This command was introduced.
12.4(6)T
The keypad-character argument was modified to allow two-digit codes.
Usage Guidelines
This command is used with the STC application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.
To use the speed-dial to voice-mail feature on a phone, dial the feature speed-dial (FSD) prefix and the code that has been configured with this command (or the default if this command was not used). For example, if the FSD prefix is * (the default), and you want to dial the voice-mail phone number, dial *0.
Note that the number that will be speed-dialed for voice mail must be set on Cisco CallManager or the Cisco CallManager Express system.
This command is reset to its default value if you modify the value of the digit command. For example, if you set the digit command to 2, then change the digit command back to its default of 1, the voice-mail FSD code is reset to 0 (zero).
If you set this code to a value that is already in use for another FSD code, you receive a warning message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command.
The show running-config command displays nondefault FSD codes only. The show stcapp feature codes command displays all FSD codes.
Examples
The following example sets an FSD prefix of two pound signs (##) and a voice-mail code of 8. After these values have been configured, a phone user presses ##8 to dial the voice-mail number.
Router(config)# stcapp feature speed-dialRouter(stcapp-fsd)# prefix ##Router(stcapp-fsd)# voicemail 8Router(stcapp-fsd)# exitRelated Commands
voiceport
To enable a private line automatic ringdown (PLAR) connection for an analog phone, use the voiceport command in SCCP PLAR configuration mode. To remove PLAR from the voice port, use the no form of this command.
voiceport port-number dial dial-string [digit dtmf-digits [wait-connect wait-msecs] [interval inter-digit-msecs]]
no voiceport port-number
Syntax Description
Command Default
Disabled (PLAR is not set for the voice port).
Command Modes
SCCP PLAR configuration
Command History
Usage Guidelines
This command enables PLAR on analog FXS ports that use Skinny Client Control Protocol (SCCP) for call control. If the digit keyword is not used, DTMF digits are not out-pulsed; the voice port uses a simple PLAR connection and the other keywords are not available.
Voice ports can be configured in any order. For example, you can configure port 2/23 before port 2/0. The show running-config command lists the ports in ascending order.
Before a PLAR port can become operational, the STC application must first be enabled in the corresponding dial-peer using the service stcapp command. If you configure a port for PLAR before enabling the STC application in the dial peer you receive a warning message.
PLAR phones support most of the same features as normal analog phones. The PLAR phone handles incoming calls and supports hookflash for basic supplementary features such as call transfer, call waiting, and conference. The PLAR phone does not support other features such as call forwarding, redial, speed dial, call park, call pick up from a PLAR phone, AMWI, or caller ID.
Examples
The following example enables the PLAR feature on port 2/0, 2/1, and 2/3. When a phone user picks up the handset on the analog phone connected to port 2/0, the system automatically rings extension 3660 and after waiting 500 milliseconds, dials 1234. The DTMF digits are out-pulsed to the destination port at an interval of 200 milliseconds.
Router(config)# sccp plarRouter(config-sccp-plar)# voiceport 2/0 dial 3660 digit 1234 wait-connect 500 interval 200Router(config-sccp-plar)# voiceport 2/1 dial 3264 digit 678,,,9*0,,#123 interval 100Router(config-sccp-plar)# voiceport 2/3 dial 3478 digit 34567 wait-connect 500Related Commands
dial-peer voice
Enters dial peer configuration mode and defines a dial peer.
sccp plar
Enters SCCP PLAR configuration mode.
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 series, 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
Syntax Description
Cisco 1750 and Cisco 1751
Cisco 2600 series, Cisco 3600 Series, and Cisco 7200 Series
Cisco AS5300:
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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
Examples
The following example accesses voice-port configuration mode for port 0, located on subunit 0 on a VIC installed in slot 1:
voice-port 1/0/0The following example accesses voice-port configuration mode for a Cisco AS5300:
voice-port 1:DRelated Commands
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.
Command Default
The voice ports on the interface are not in busyout state.
Command Modes
Interface configuration
Command History
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 busyoutThe following example places the voice ports associated with ATM interface 0 into busyout state:
interface atm 0voice-port busyoutRelated Commands
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.
Command Default
The voice path is cut-through in only the backward direction when the RTP channel is opened.
Command Modes
Global configuration
Command History
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 rtp send-recvvoice 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.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 service voatmvoice 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 of the source IP group name is 31 alphanumeric characters.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 utah2Related Commands
voice statistics accounting method
To enable voice accounting statistics to be collected for a specific accounting method list and to specify the pass criteria for call legs, use the voice statistics accounting method command in global configuration mode. To disable the collection of statistics for the accounting method, use the no form of this command.
voice statistics accounting method method-list-name pass {start-interim-stop | start-stop | stop-only}
no voice statistics accounting method method-list-name pass {start-interim-stop | start-stop | stop-only}
Syntax Description
Command Default
No statistics for the specified accounting method list are collected.
Command Modes
Global configuration
Command History
Examples
The following example shows that h323 is specified as the method list and that the pass criterion is stop-only:
Router(config)# voice statistics accounting method h323 pass stop-onlyRelated Commands
voice statistics display-format separator
To configure the display format of the statistics on the gateway, use the voice statistics display-format separator command in global configuration mode. To return the display format of the statistics to the default value, use the no form of this command.
voice statistics display-format separator {space | tab | new-line | char char}
no voice statistics display-format separator {space | tab | new-line | char char}
Syntax Description
Command Default
A comma (,) is the default separator.
Command Modes
Global configuration
Command History
Examples
The following example shows that a space is specified as the display separator:
Router(config)# voice statistics display-format separator spaceRelated Commands
voice statistics field-params
To configure the parameters of call statistics fields on the gateway, use the voice statistics field-params command in global configuration mode. To return the call statistics parameters to the default values, use the no form of this command.
voice statistics field-params {mcd value | lost-packet value | packet-latency value | packet-jitter value}
no voice statistics field-params {mcd value | lost-packet value | packet-latency value | packet-jitter value}
Syntax Description
Command Default
MCD is 2 milliseconds.
Lost packet threshold is 1000 milliseconds.
Packet latency threshold is 250 milliseconds.
Packet jitter threshold is 250 milliseconds.Command Modes
Global configuration
Command History
Examples
The following example configures a minimum call duration of 5 milliseconds:
Router(config)# voice statistics field-params mcd 5The following example configures a lost packet threshold of 250 milliseconds:
Router(config)# voice statistics field-params lost-packet 250The following example configures a packet-latency threshold of 300 milliseconds:
Router(config)# voice statistics field-params packet-latency 300The following example configures a packet-jitter threshold of 245 milliseconds:
Router(config)# voice statistics field-params packet-jitter 245Related Commands
voice statistics max-storage-duration
To configure the maximum amount of time for which collected statistics are stored in the system memory of the gateway, use the voice statistics max-storage-duration command in global configuration mode. To remove the configured maximum storage duration, use the no form of this command.
voice statistics max-storage-duration {day value | hour value | minute value}
no voice statistics max-storage-duration {day value | hour value | minute value}
Syntax Description
Command Default
If no length of time is configured, no memory is allocated for those call statistic records that have stopped after the end of their collection intervals. If no memory is allocated, only active call statistic record buffers are kept in system memory.
Command Modes
Global configuration
Command History
Usage Guidelines
The maximum storage duration means the time-to-exist duration of the call statistic records on the gateway.
The values entered using this command also apply to the collection of VoIP internal error codes (IECs).
Examples
The following example shows that the maximum storage duration for the collection of voice call statistics has been set for 60 minutes:
Router(config)# voice statistics max-storage-duration minute 60Related Commands
voice statistics push
To configure the method for pushing signaling statistics, VoIP AAA accounting statistics, or Cisco internal error codes (IECs) to an FTP or syslog server, use the voice statistics push command in global configuration mode. To disable the configured push method, use the no form of this command.
voice statistics push {ftp url ftp-url [max-file-size value]} | {syslog [max-msg-size value]}
no voice statistics push {ftp url ftp-url [max-file-size value]} | {syslog [max-msg-size value]}
Syntax Description
Command Default
Voice statistics are not pushed to an FTP or syslog server.
Command Modes
Global configuration
Command History
Usage Guidelines
The gateway configuration should be consistent with the configuration on the FTP or syslog servers. This command may also be used to push Cisco VoIP internal error codes (IECs) to either an FTP server or a syslog server.
Examples
The following is a configuration example showing a specified FTP server and maximum file size:
Router(config)# voice statistics push ftp url ftp://john:doe@abc:23//directory1/directory2 max-file-size 10000Related Commands
voice statistics time-range
To specify a time range to collect statistics from the gateway on a periodic basis, since the last reset, or for a specific time duration , use the voice statistics time-range command in global configuration mode. To disable the time-range settings, use the no form of this command.
Statistics Collection on a Periodic Basis
voice statistics time-range periodic interval start hh:mm {days-of-week {Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday | daily | weekdays | weekend}} [end hh:mm {days-of-week {Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday}}]
no voice statistics time-range periodic interval start hh:mm {days-of-week {Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday | daily | weekdays | weekend}} [end hh:mm {days-of-week {Monday | Tuesday | Wednesday | Thursday | Friday | Saturday | Sunday}}]
Statistics Collection Since the Last Reset or Reboot of the Gateway
voice statistics time-range since-reset
no voice statistics time-range since-reset
Statistics Collection at a Specific Time Duration
voice statistics time-range specific start hh:mm day month year end hh:mm day month year
no voice statistics time-range specific start hh:mm day month year end hh:mm day month year
Syntax Description
No statistics are collected by default.
Command Modes
Global configuration
Command History
Usage Guidelines
There should be only one specific or periodic configuration at any one time. If a second specific or periodic configuration is configured, the request is rejected and a warning message displays. If the no form of the command is used during the specific time range, the corresponding collection will stop and FTP or syslog messages will not be sent.
Examples
The following example shows that the time range is periodic and set to collect statistics for a 60-minute period on weekdays only beginning at 12:00 a.m.:
Router(config)# voice statistics time-range periodic 60minutes start 12:00 days-of-week weekdaysThe following example configures the gateway to collect call statistics since the last reset (specified with the clear voice statistics csr command) or since the last time the gateway was rebooted:
Router(config)# voice statistics time-range since-resetThe following example configures the gateway to collect statistics from 10:00 a.m. on the first day of January to 12:00 a.m. on the second day of January:
Router(config)# voice statistics time-range specific start 10:00 1 January 2004 end 12:00 2 January 2004Related Commands
voice statistics type csr
To configure a gateway to collect VoIP AAA accounting statistics or voice signaling statistics, independently or at the same time, use the voice statistics type csr command in global configuration mode. To disable the counters, use the no form of this command.
voice statistics type csr [accounting | signaling]
no voice statistics type csr [accounting | signaling]
Syntax Description
accounting
(Optional) VoIP AAA accounting statistics are collected.
signaling
(Optional) Voice signaling statistics are collected.
Command Default
No accounting or signaling call statistics records (CSRs) are collected on the gateway.
Command Modes
Global configuration
Command History
Usage Guidelines
If you do not specify a keyword, both accounting and signaling CSRs are collected. Accounting and signaling CSR collection can be enabled and disabled independently.
Examples
The following example shows that both types of CSRs will be collected:
Router(config)# voice statistics type csrThe following example enables accounting CSRs to be collected:
Router(config)# voice statistics type csr accountingThe following example enables signaling CSRs to be collected:
Router(config)# voice statistics type csr signalingThe following example disables the collection of both signaling and accounting CSRs:
Router(config)# no voice statistics type csrThe following example disables the collection of signaling CSRs only:
Router(config)# no voice statistics type csr signalingRelated Commands
voice statistics type iec
To enable collection of Internal Error Code (IEC) statistics, use the voice statistics type iec command in global configuration mode. To disable IEC statistics collection, use the no form of this command.
voice statistics type iec
no voice statistics type iec
Syntax Description
This command has no arguments or keywords.
Command Default
IEC statistics collection is disabled.
Command Modes
Global configuration
Command History
Examples
The following example enables IEC statistics collection:
router(config)# voice statistics type iecRelated Commands
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 of the voice translation profile name is 31 alphanumeric characters.
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 westcoastRouter(cfg-translation-profile)# translate calling 2Router(cfg-translation-profile)# translate called 1Router(cfg-translation-profile)# translate redirect-called 3Related Commands
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
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 150Router(cfg-translation-rule)# rule 1 reject /^408\(.(\)/Router(cfg-translation-rule)# rule 2 /\(^...\)853\(...\)/ /\1525\2/Related Commands
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. The default is 250.
Command Default
250 milliseconds
Command Modes
Global configuration
Command History
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 Cisco IOS Release 12.1(2)T.
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:
voice vad-time 20000Related Commands
voice vrf
To configure a voice VRF, use the voice vrf command in global configuration mode. To remove the voice VRF configuration, use the no form of this command.
voice vrf vrfname
no voice vrf vrfname
Syntax Description
Command Default
No voice VRF is configured.
Command Modes
Global configuration
Command History
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
Usage Guidelines
You must create a VRF using the ip vrf vrfname command before you can configure it as a voice VRF.
To ensure there are no active calls on the voice gateway during a VRF change, voices services must be shut down on the voice gateway before you configure or make changes to a voice VRF.
Examples
The following example shows that a VRF called vrf1 was created and then configured as a voice VRF:
ip vrf vrf1rd 1:1route-target export 1:2route-target import 1:2!voice vrf vrf1!voice service voipRelated Commands
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
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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-definitionRelated Commands
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 {calling | called} name-tag
no voip-incoming translation-rule {calling | called} name-tag
Syntax Description
Command Default
No default behavior or values
Command Modes
Global configuration
Command History
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 called 5Related Commands
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
A number from 1 to 5 representing decibels (dB) of gain. Range is as follows:
•
•
•
•
•
Default is 3 (-7.7 dB gain).
Command Default
3 (-7.7 dB gain)
Command Modes
Dial peer voice configuration
Command History
12.2(8)T
This command was introduced on Cisco 803, Cisco 804, and Cisco 813 routers.
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
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.
dial-peer voice 1 potsdestination-pattern 5551111port 1no call-waitingring 0volume 4dial-peer voice 2 potsdestination-pattern 5552222port 2no call-waitingring 0volume 2Related Commands
show pots volume
Shows the receiver volume configured for each POTS port on a router.
vxml audioerror
To enable throwing an error event when audio playout fails, use the vxml audioerror command in global configuration mode. To return to the default, use the no form of this command.
vxml audioerror
no vxml audioerror
Syntax Description
This command has no arguments or keywords.
Command Default
An audio error event, error.badfetch, is not thrown when an audio file cannot be played.
Command Modes
Global configuration
Command History
Usage Guidelines
Entering this command causes an audio error event, error.badfetch, to be thrown when an audio file cannot be played, for instance, because the file is in an unsupported format, the src attribute references an invalid URI, or the expr attribute evaluates to an invalid URI.
The vxml audioerror command overrides the vxml version 2.0 command, so that if both commands are entered, the audio error event will be thrown when an audio file cannot be played.
Examples
The following example enables the audio error feature:
Router(config)# vxml audioerrorRelated Commands
vxml version pre2.0
Enables features compatible with versions earlier than VoiceXML 2.0.
vxml tree memory
To set the maximum memory size for the VoiceXML parser tree, use the vxml tree memory command in global configuration mode. To reset to the default, use the no form of this command.
vxml tree memory size
no vxml tree memory
Syntax Description
Defaults
1000 KB
Command Modes
Global configuration
Command History
12.2(15)T
This command was introduced.
12.4(15)T
The default was changed from 64 to 1000.
Usage Guidelines
This command limits the memory resources available for parsing VoiceXML documents, preventing large documents from consuming excessive system memory. Increasing the maximum memory size for the VoiceXML tree enables calls to use larger VoiceXML documents. If a VoiceXML document exceeds the limit, the gateway aborts the document execution and the debug vxml error command displays a "vxml malloc fail" error.
Note
Examples
The following example sets the maximum memory size to 128 KB:
vxml tree memory 128Related Commands
vxml version 2.0
To enable VoiceXML 2.0 features, use the vxml version 2.0 command in global configuration mode. To return to the default, use the no form of this command.
vxml version 2.0
no vxml version 2.0
Syntax Description
This command has no arguments or keywords.
Command Default
The default VoiceXML behavior is compatible with versions earlier than W3C VoiceXML 2.0 Specification.
Command Modes
Global configuration
Command History
Usage Guidelines
This command enables the following VoiceXML features:
•
•
•
•
Examples
The following example enables VoiceXML version 2.0 features:
Router(config)# vxml version 2.0
