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Cisco IOS Software Releases 12.0 Special and Early Deployments

New and Changed Commands in Cisco IOS Release 12.0(7)XK

Table Of Contents

New and Changed Commands in Cisco IOS Release 12.0(7)XK

Related Documents

Command Reference

auto-cut-through

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

battery-reversal

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

ccs connect

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

ccs encap frf11

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

codec (dial-peer)

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

Related Commands

codec (voice-port)

codec complexity

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Command

condition

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

connect

connect voice

connection

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

define

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

dial-peer hunt

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

Related Commands

dial-peer terminator

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

Related Commands

dial-peer voice

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

disconnect-ack

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

ds0-group

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

encapsulation

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

encapsulation ftc-trunk

forward-digits

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

frag-pre-queuing

frame-relay interface-dlci

Syntax Description

Defaults

Command Mode

Command History

ftc-trunk frame-relay-dlci

ftc-trunk management-dlci

ftc-trunk management-protocol

ftc-trunk voice-dlci

huntstop

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

icpif

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

ignore

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

incoming called-number

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

isdn contiguous-bchan

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

Related Commands

isdn incoming-voice

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

isdn protocol-emulate

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

isdn switch type

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

loss-plan

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

Related Commands

num-exp

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Example

Related Commands

playout delay

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

pri-group

Syntax Description

Default

Command Mode

Command History

Usage Guidelines

Example

Related Command

ring cadence

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

session target

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

show call active voice

Syntax Description

Command Mode

Command History

Usage Guidelines

Example

Related Commands

show call history voice

Syntax Description

Command Mode

Command History

Usage Guidelines

Example

Related Commands

show num-exp

Syntax Description

Command Mode

Command History

Usage Guidelines

Example

Related Commands

show voice call

Syntax Description

Command Mode

Command History

Usage Guidelines

Sample Display

Related Commands

show voice dsp

Syntax Description

Command Mode

Command History

Usage Guidelines

Sample Display

Related Commands

show voice port

Syntax Description

Command Mode

Command History

Usage Guidelines

Example

Related Commands

show voice trunk-conditioning signaling

Syntax Description

Command Mode

Command History

Usage Guidelines

Sample Display

Related Commands

show voice trunk-conditioning supervisory

Syntax Description

Command Mode

Command History

Usage Guidelines

Sample Display

Related Commands

signal pattern

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

signal sequence oos

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

signal timing idle suppress-voice

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

signal-type

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

test voice port detector

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

test voice port inject-tone

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

test voice port loopback

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

test voice port relay

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

test voice port switch

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

timeouts ringing

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

timeouts wait-release

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

timing guard-out

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

timing percentbreak

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

vbr-rt

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

vofr

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

voice-card

Syntax Description

Command Mode

Command History

Usage Guidelines

Example

Related Commands

voice-encap

voice-group

voice local-bypass

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

voice vad-time

Syntax Description

Defaults

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

Debug Commands

debug ccfrf11 session

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

debug ccswvoice voatm-debug

Syntax Description

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccswvoice voatm-session

Syntax Description

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccswvoice vofr-debug

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

debug ccswvoice vofr-session

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands

debug voice all

debug voice cp

debug voice eecm

debug voice protocol

debug voice signaling

debug voice vofr

debug vpm all

Syntax Description

Defaults

Command History

Usage Guidelines

Examples

Related Commands

debug vpm trunk_sc

Syntax Description

Defaults

Command History

Usage Guidelines

Examples

Related Commands

debug vtsp all

Syntax Description

Defaults

Command History

Usage Guidelines

Related Commands

debug vtsp dsp

Syntax Description

Defaults

Command History

Usage Guidelines

Sample Display

Related Commands

debug vtsp error

Syntax Description

Defaults

Command History

Usage Guidelines

Sample Display

Related Commands

debug vtsp port

Syntax Description

Defaults

Command History

Usage Guidelines

Examples

Related Commands

debug vtsp session

Syntax Description

Defaults

Command History

Usage Guidelines

Sample Display

Related Commands

debug vtsp stats

Syntax Description

Defaults

Command History

Usage Guidelines

Sample Display

Related Commands

debug vtsp vofr subframe

Syntax Description

Command Mode

Command History

Usage Guidelines

Examples

Related Commands


New and Changed Commands in Cisco IOS Release 12.0(7)XK


This document provides a consolidated command reference of all the new, changed, and removed commands in Cisco IOS Release 12.0(7)XK.

This document includes the following sections:

Command Reference

Debug Commands

Related Documents

The command reference entries in this document are also included in one or more of the following 12.0(7)XK online documents:

Configuring Voice over IP on Cisco MC3810 Concentrators

Voice over ATM on the Cisco 3600 Series Routers

Voice over Frame Relay Using FRF.11 and FRF.12 Configuration Updates

Voice Port Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators

Voice Port Testing Enhancements on Cisco 2600, 3600 and MC3810

Configuring Cisco MC3810 Series Concentrators to Use High-Performance Compression Modules

QSIG Protocol Support on Cisco MC3810, 7200, 2600, and 3600 Series Routers

Transparent CCS and Frame Forwarding Enhancements on the Cisco MC3810

Trunk Conditioning Enhancements in MC3810 Series Concentrators

Voice Busyout Enhancements on the Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators

Configuring Digital E1 Packet Voice Trunk Network Module Interfaces

Command Reference

This section documents new, modified and removed commands. Modified commands are indicated by an asterisk (*). All other commands used on these platforms are documented in the Cisco IOS Release 12.0 command reference publications.

auto-cut-through*

battery-reversal

ccs connect

ccs encap frf11

codec (dial-peer)*

codec complexity*

condition*

connection*

define*

dial-peer hunt

dial-peer terminator*

dial-peer voice*

disconnect-ack*

ds0-group*

encapsulation*

encapsulation ftc-trunk*

forward-digits*

frame-relay interface-dlci*

huntstop*

icpif*

ignore*

incoming called-number*

isdn contiguous-bchan

isdn incoming-voice*

isdn protocol-emulate*

isdn switch type*

loss-plan*

num-exp*

playout delay*

pri-group*

ring cadence*

session target*

show call active voice*

show call history voice*

show num-exp*

show voice call*

show voice dsp*

show voice port*

show voice trunk-conditioning signaling

show voice trunk-conditioning supervisory

signal pattern*

signal sequence oos

signal timing idle suppress-voice

signal-type*

test voice port detector

test voice port inject-tone

test voice port loopback

test voice port relay

test voice port switch

timeouts ringing

timeouts wait-release*

timing guard-out*

timing percentbreak*

vbr-rt*

vofr*

voice-card*

voice local-bypass

voice vad-time

The following commands have been removed in Cisco IOS Release 12.0(7)XK:

codec (voice-port)*

connect

connect voice

frag-pre-queuing

ftc-trunk frame-relay-dlci

ftc-trunk management-dlci

ftc-trunk management-protocol

ftc-trunk voice-dlci

voice-encap

voice-group

auto-cut-through

To enable call completion when a PBX does not provide an M-lead response, use the auto-cut-through voice-port configuration command. Use the no form of this command to disable the auto-cut-through operation.

auto-cut-through
no auto-cut-through

Syntax Description

This command has no arguments or keywords.

Defaults

Auto-cut-through is enabled.

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

The auto-cut-through command applies to E&M voice ports only.

Examples

The following example enables call completion on a Cisco MC3810 when a PBX does not provide an M-lead response:

router(config)# voice-port 1/1
router(config-voiceport)# auto-cut-through

The following example enables call completion on a Cisco 2600 or 3600 when a PBX does not provide an M-lead response:

router(config)# voice-port 1/0/0
router(config-voiceport)# auto-cut-through

Related Commands

Command
Description

show voice port

Displays voice port configuration information.


battery-reversal

To specify battery polarity reversal on an FXO or FXS port, use the battery-reversal voice-port configuration command. Use the no form of this command to disable battery reversal.

battery-reversal
no battery-reversal

Syntax Description

This command has no arguments or keywords.

Defaults

Battery reversal is enabled.

Command Mode

Voice-port configuration

Command History

Release
Modification

12.0(7)XK

This command was introduced.


Usage Guidelines

The battery-reversal command applies to FXO and FXS voice ports. On Cisco 2600 and 3600 series routers, only analog voice ports in VIC-2FXO-M1 and VIC-2FXO-M2 voice interface cards are able to detect battery reversal; analog voice ports in VIC-2FXO and VIC-2FXO-EU voice interface cards do not detect battery reversal. On digital voice ports, battery reversal is only supported on E1 MELCAS; it is not supported in T1 channel associated signaling (CAS) or E1 CAS.

FXS ports normally reverse battery upon call connection. If an FXS port is connected to an FXO port that does not support battery reversal detection, you can use the no battery-reversal command on the FXS port to prevent unexpected behavior.

FXO ports in loopstart mode normally disconnect calls when they detect a second battery reversal (back to normal). You can use the no battery-reversal command on FXO ports to disable this action.

The battery-reversal command restores voice ports to their default battery-reversal operation.

Examples

The following example disables battery reversal on voice port 1/1 on a Cisco MC3810:

router(config)# voice-port 1/1
router(config-voiceport)# no battery-reversal

The following example disables battery reversal on voice port 1/0/0 on a Cisco 2600 or 3600 series router:

router(config)# voice-port 1/0/0
router(config-voiceport)# no battery-reversal

Related Commands

Command
Description

show voice port

Displays voice port configuration information.


ccs connect

To configure a CCS connection on an interface configured to support CCS frame forwarding, use the ccs connect interface configuration command. To disable the CCS connection on the interface, use the no form of this command.

ccs connect {serial | atm} number [ dlci | pvc vpi/vci | pvc name ] [ cidnumber ]
no ccs connect {serial | atm} number [ dlci | pvc vpi/vci | pvc name ] [ cidnumber ]

Syntax Description

number

Specify the connection number.


The following parameters are used for Frame Relay configuration:

serial

Make a serial CCS connection.

dlci

Specify the DLCI number.

cidnumber

(Optional) If you have executed the ccs encap frf11 command, the cid option allows you to specify any CID number from 5 to 255.


atm

Make an ATM CCS connection.

pvc vpi/vci

Specify the PVC virtual path identifier/virtual channel identifier. Acceptable values are from 0 to 255; the slash is required.

pvc name

Specify the PVC string that names the PVC for recognition.


The following parameters are used for ATM configuration:

Defaults

No CCS connection is made.

Command Mode

Serial interface configuration mode

Command History

Release
Modification

12.0(2)T

This command was introduced for the Cisco MC3810.

12.0(7)XK

Added CID syntax, removed dlci keyword and vcd options.


Usage Guidelines

Use this command to configure a CCS connection. If the CCS connection is over Frame Relay, specify a serial interface and the DLCI. If the CCS connection is over ATM, specify atm, the interface number (0 only on the Cisco MC3810), and the PVC.

If you have executed the ccs encap frf11 command, the cidnumber option allows you to specify any CID from 5 to 255. If you do not issue the ccs encap frf11 command, Cisco encapsulation is used, and any CID value other than 254 is ignored.


Note   CDP and keepalives are disabled by default on a D channel interface.


Examples

To configure a frame relay CCS frame-forwarding connection on DLCI 100 by using the default CID of 254, enter the following command:

ccs connect serial 1 100

or:

ccs connect serial 1 100 10

To configure a CCS frame-forwarding connection over an ATM PVC, enter the following command:

ccs connect atm0 pvc 100/10

or:

ccs connect atm0 pvc 10/100 21

or:

ccs connect atm0 pvc mypvc_10 21

To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 using a CID of 110, enter the following command:

ccs connect serial 1 100 110

Related Commands

Command
Description

ccs encap frf11

Allows the specification of the standard Annex-C FRF.11 format.


ccs encap frf11

To configure the common channel signaling (CCS) packet encapsulation format for FRF.11, use the ccs encap frf11 command. Use the no form of this command to disable ccs encapsulation for FRF11.

ccs encap frf11

no ccs encap frf11

Syntax Description

There are no keywords or arguments.

Defaults

By default, the format is a Cisco packet format, using a channel ID (CID) of 254.

Command Mode

Serial configuration mode

Command History

Release
Modification

12.0(7)XK

This command was introduced for the Cisco MC3810.


Usage Guidelines

This command allows the specification of the standard Annex-C format. Use this command to define the packet format for the CCS packet; it places the FRF.11 Annex-C (Data Transfer Syntax) standard header on the CCS packets only.

Once the ccs encap frf11 command is executed, you can use the ccs connect command to specify a CID other than 254.

Examples

The following example shows how to configure a serial interface for Frame Relay:

router(config)# interface Serial1:15
router(config-if)# ccs encap frf11
router(config-if)# ccs connect Serial0 990 100

Related Commands

Command
Description

mode ccs frame-forwarding

Set to forward frames on the controller.


codec (dial-peer)

To specify the voice codec for a network dial peer, enter the codec dial-peer configuration command. Use the no form of this command to restore the default value.

codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8}[bytes payload-size]
no codec

Syntax Description

codec

Codec options on the Cisco MC3810 with codec complexity set to high or medium:

g711alaw—G.711 A Law, 64000 bps

g711ulaw—G.711 u Law, 64000 bps

g723ar53—G.723.1 Annex A, 5300 bps

g723ar63—G.723.1 Annex A, 6300 bps

g723r53—G.723.1, 5300 bps

g723r63—G.723.1, 6300 bps

g726r16—G.726, 16000 bps

g726r24—G.726, 24000 bps

g726r32—G.726, 32000 bps

g728—G.728, 16000 bps

g729abr8—G.729 Annex A and Annex B, 8000 bps

g729ar8—G.729 Annex A, 8000 bps

g729br8—G.729 Annex B, 8000 bps

g729r8—G.729, 8000 bps

bytes

(Optional) The voice payload for each frame.

payload-size

(Optional) Number of bytes you specify as the voice payload of each frame. Acceptable values are from 10 to 240 in increments of 10 (10, 20, 30 ... 220, 230, 240). Any other value is rounded down.


Defaults

Dial peers are configured for g729r8.

Command Mode

Dial-peer configuration

Command History

Release
Modification

11.3(1)T

This command was introduced as a Cisco 3600 VoIP dial-peer configuration command.

12.0(4)T

This command was modified for VoFR dial peers. On the Cisco MC3810, this command was first supported as a dial-peer command.

12.0(5)XK and 12.0(7)T

The g729br8 codec and pre-ietf keyword were added for the Cisco 2600 and 3600 platforms.

12.0(7)XK

The g729abr8 and g729ar8 codecs were added for the Cisco MC3810 and the keyword pre-ietf was deleted.


Usage Guidelines

A codec type can be configured on the dial-peer if it is supported under the codec complexity setting you have specified.

The dial-peer configuration command is particularly useful when you must change to a small-bandwidth codec. Large-bandwidth codecs, such as G.711, do not fit in a small-bandwidth link. However, g711alaw and g711ulaw provide higher-quality voice transmission than other codecs. For almost toll quality (and a significant savings in bandwidth), g729r8 provides near-toll quality with considerable bandwidth savings.

If the destination router does not support a codec required by the originating router, the call setup fails.

You can change the payload of each voice packet frame by using the bytes payload-size setting. However, increasing the payload size can add processing delay for each voice packet. Table 1 describes the voice payload options and default values for the codecs and packet voice protocols.

Table 1 Voice Payload-per-Frame Options and Defaults 

Codec
Protocol
Voice Payload Options (bytes)
Default Voice Payload (bytes)

g711alaw
g711ulaw

VoIP
VoFR
VoATM

80, 160
40 to 240 in multiples of 40
40 to 240 in multiples of 40

160
240
240

g723ar53
g723r53

VoIP
VoFR
VoATM

20 to 220 in multiples of 20
20 to 240 in multiples of 20
20 to 240 in multiples of 20

20
20
20

g723ar63
g723r63

VoIP
VoFR
VoATM

24 to 216 in multiples of 24
24 to 240 in multiples of 24
24 to 240 in multiples of 24

24
24
24

g726r16

VoIP
VoFR
VoATM

20 to 220 in multiples of 20
10 to 240 in multiples of 10
10 to 240 in multiples of 10

40
60
60

g726r24

VoIP
VoFR
VoATM

30 to 210 in multiples of 30
15 to 240 in multiples of 15
30 to 240 in multiples of 15

60
90
90

g726r32

VoIP
VoFR
VoATM

40 to 200 in multiples of 40
20 to 240 in multiples of 20
40 to 240 in multiples of 20

80
120
120

g728

VoIP
VoFR
VoATM

10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10

40
60
60

g729abr8
g729ar8
g729br8
g729r8

VoIP
VoFR
VoATM

10 to 230 in multiples of 10
10 to 240 in multiples of 10
10 to 240 in multiples of 10

20
30
30


Example

The following example configures VoIP dial peer number 10 to use codec type g723r53 (G.723.1 at 5300 bps):

router(config)# dial-peer voice 10 voip 
router(config-dialpeer)# codec g723r53

Related Commands

Command
Description

codec complexity

This voice-card configuration command sets codec complexity and call density.

show dial-peer voice

Displays the codec setting for dial peers.


codec (voice-port)

The codec voice-port configuration command on the Cisco MC3810 is no longer supported beginning in this release. This command was first supported in Cisco IOS Release 11.3(1)MA. Configure the codec value using the codec dial-peer configuration command.

codec complexity

To match the DSP complexity packaging to the codec(s) to be supported, enter the codec complexity voice-card configuration command. The no form of the command restores the default value.

codec complexity {high | medium}
no codec complexity

Syntax Description

high

With high complexity packaging, each DSP supports two voice channels encoded in any of the following formats: G.711ulaw, G.711alaw, G.723.1(r5.3), G.723.1 Annex A(r5.3), G.723.1(r6.3), G.723.1 Annex A(r6.3), G.726(r16), G.726(r24), G.726(r32), G.729, G.729 Annex B, G.728, and fax relay.

medium

With medium complexity packaging, each DSP supports four voice channels encoded in any of the following formats: G.711ulaw, G.711alaw, G.726(r16), G.726(r24), G.726(r32), G.729 Annex A, G.729 Annex B with Annex A, and fax relay. This is the default.


Defaults

The default is medium complexity.

Command Mode

Voice-card configuration

Command History

Release
Modification

12.0(5)XK and 12.0(7)T

The command was introduced for the Cisco 2600 and 3600 series.

12.0(7)XK

This command was first supported on the Cisco MC3810 platform for use with the high performance compression module (HCM).


Usage Guidelines

Select a higher codec complexity if that is required in order to support a particular codec or combination of codecs.

Select a lower codec complexity to support the greatest number of voice channels, provided that the lower complexity is compatible with the particular codecs in use.

To change codec complexity, all of the DSP voice channels must be in the idle state.

Codec complexity refers to the amount of processing required to perform voice compression. Codec complexity affects the call density—the number of calls that can take place on the digital signal processors (DSPs). With higher codec complexity, fewer calls can be handled.


Note   On the Cisco MC3810, this command is valid only with HCM(s) installed, and you must specify voice card 0 in the command mode. If two HCMs are installed, the codec complexity command configures both HCMs at once.


Examples

The following example sets the codec complexity to high on a Cisco MC3810 containing one or two HCMs:

router(config)# voice-card 0 
router(config-voicecard)# codec complexity high

The following example sets the codec complexity to high on voice card 1 in a Cisco 2600 or 3600 router:

router(config)# voice-card 1 
router(config-voicecard)# codec complexity high

Related Command

Command
Description

show voice dsp

Shows the current status of all DSP voice channels.


condition

To manipulate the signaling format bit-pattern for all voice signaling types, use the condition command. Use the no form of this command to turn off conditioning on the voice port.

condition {tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit} {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit} {on | off | invert}
no condition {tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit} {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit} {on | off | invert}

Syntax Description

tx-a-bit

Transmit A bit.

tx-b-bit

Transmit B bit.

tx-c-bit

Transmit C bit.

tx-d-bit

Transmit D bit.

rx-a-bit

Receive A bit.

rx-b-bit

Receive B bit.

rx-c-bit

Receive C bit.

rx-d-bit

Receive D bit.

on

Forces the bit state to be 1.

off

Forces the bit state to be 0.

invert

Inverts the bit state.


Defaults

The signaling format is not manipulated (for all transmit or receive A, B, C, and D bits).

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

Use the condition command to manipulate the sent or received bit patterns to match expected patterns on a connected device. Be careful not to destroy the information content of the bit pattern. For example, forcing the A-bit on or off will prevent FXO interfaces from being able to generate both an on-hook and off-hook state.

Examples

The following example manipulates the signaling format bit-pattern on voice port 1/1 on a Cisco MC3810:

router(config)# voice-port 1/1
router(config-voiceport)# condition tx-a-bit invert
router(config-voiceport)# condition rx-a-bit invert

The following example manipulates the signaling format bit-pattern on voice port 1/1/2 on a Cisco 2600 or 3600:

router(config)# voice-port 1/0/0
router(config-voiceport)# condition tx-a-bit invert
router(config-voiceport)# condition rx-a-bit invert

Related Commands

Command
Description

define

Defines the transmit and receive bits for E&M and E&M MELCAS voice signaling.

ignore

Configures the E&M or E&M MELCAS voice port to ignore specific receive bits.


connect

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

connect voice

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

connection

To specify a connection mode for a voice port, use the connection voice-port configuration command. Use the no form of this command to disable the selected connection mode.

connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}
no connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}

Syntax Description

plar

Specifies a private line automatic ring down (PLAR) connection. PLAR is an autodialing mechanism that permanently associates a voice interface with a far-end voice interface, allowing call completion to a specific telephone number or PBX without dialing. When the calling telephone goes off hook a predefined network dial peer is automatically matched, which sets up a call to the destination telephone or PBX.

tie-line

Specifies a connection that emulates a temporary tie-line trunk to a private branch exchange (PBX). A tie-line connection is automatically set up for each call and torn down when the call ends.

plar-opx

Specifies a PLAR Off-Premises eXtension connection. Using this option, the local voice port provides a local response before the remote voice port receives an answer. On FXO interfaces, the voice port will not answer until the remote side answers.

trunk

Specifies a connection that emulates a permanent trunk connection to a private branch exchange (PBX). A trunk connection remains "nailed up" in the absence of any active calls.

digits

Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number.

answer-mode

(Optional; used only with the trunk keyword.) Specifies that the router should not attempt to initiate a trunk connection, but should wait for an incoming call before establishing the trunk.


Defaults

No connection mode is specified.

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3(1)T

This command was introduced.

11.3(1)MA1

This command was first supported on the Cisco MC3810, and the tie-line keyword was first made available on the Cisco MC3810.

11.3(1)MA5 and 12.0(2)T

The plar-opx keyword was first made available on the Cisco MC3810 as the plar-opx-ringrelay keyword. The keyword was shortened in a subsequent release.

12.0(3)XG

The trunk keyword was made available on the Cisco MC3810.

The trunk answer-mode option was added.

12.0(7)XK

This command was unified across the Cisco 2600, 3600, and MC3810 platforms.


Usage Guidelines

Use this command to specify a connection mode for a specific interface. For example, use the connection plar command to specify a PLAR interface. The string you configure for this command is used as the called number for all incoming calls over this connection. The destination peer is determined by the called number.

Use the connection trunk command to specify a permanent, "nailed up" tie-line connection to a PBX. You can use the connection trunk command for E&M-to-E&M trunks, FXO-to-FXS trunks, and FXS-to-FXS trunks. Signaling will be transported for E&M-to-E&M trunks and FXO-to-FXS trunks; signaling will not be transported for FXS-to-FXS trunks.

To configure one of the devices in the trunk connection to act as slave and only receive calls, use the answer-mode option with the connection trunk command when configuring that device.


Note   When using the connection trunk command, you must perform a shutdown/no shutdown command sequence on the voice port.


Use the connection tie-line command when the dial plan requires that additional digits be added in front of any digits dialed by the PBX, and that the combined set of digits be used to route the call onto the network. The operation is similar to the connection plar command operation, but in this case the tie-line port waits to collect digits from the PBX. The tie-line digits are automatically stripped by a terminating port.

If the connection command is not configured, the standard session application outputs a dial tone when the interface goes off-hook until enough digits are collected to match a dial-peer and complete the call.

Examples

The following example selects PLAR as the connection mode on a Cisco 3600, with a destination telephone number of 555-9262:

router(config)# voice-port 1/0/0
router(config-voiceport)# connection trunk 5559262

The following example selects tie-line as the connection mode on a Cisco MC3810, with a destination telephone number of 555-9262:

router(config)# voice-port 1/1
router(config-voiceport)# connection tie-line 5559262

The following example specifies a PLAR off-premises extension connection on a Cisco 3600, with a destination telephone number of 555-9262:

router(config)# voice-port 1/0/0
router(config-voiceport)# connection plar-opx 5559262

The following example configures a Cisco 3600 series router for a trunk connection and specifies that it will establish the trunk only when it receives an incoming call:

router(config)# voice-port 1/0/0
router(config-voiceport)# connection trunk 5559262 answer-mode

Related Commands

Command
Description

session-protocol

Establishes a session protocol for calls between the local and remote routers via the packet network.

session-target

Configures a network-specific address for a dial peer.

dial-peer voice

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

destination-pattern

Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.


define

To define the transmit and receive bits for E&M and E&M Mercury Exchange Limited (MELCAS) voice signaling, use the define voice-port configuration command. Use the no form of this command to restore the default value.

define {Tx-bits | Rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 |
  0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
no define {Tx-bits | Rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 |
  0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}

Syntax Description

Tx-bits

Transmit signaling bits.

Rx-bits

Receive signaling bits.

seize

The bit pattern defines the seized state.

idle

The bit pattern defines the idle state.

0000 through 1111

Specifies the bit pattern.


Defaults

The default is to use the preset signaling patterns as defined in ANSI and CEPT standards, as follows:

For E&M:

Tx-bits idle 0000 (0001 if on E1 trunk)
Tx-bits seize 1111
Rx-bits idle 0000
Rx-bits seize 1111

For E&M MELCAS:

Tx-bits idle 1101
Tx-bits seize 0101
Rx-bits idle 1101
Rx-bits seize 0101

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3(1) MA3

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

This command applies to E&M digital voice ports associated with T1/E1 controllers.

Use the define command to match the E&M bit patterns with the attached telephony device. Be careful not to define invalid configurations, such as all 0000 on E1, or identical seized and idle states. Use this command with the ignore command.

Examples

To configure a voice port on a Cisco 2600 or 3600 router sending traffic in North American E&M signaling format to convert the signaling to MELCAS format, enter the following commands:

router(config)# voice-port 1/0/0
router(config-voiceport)# define rx-bits idle 1101
router(config-voiceport)# define rx-bits idle 0101
router(config-voiceport)# define tx-bits seize 1101
router(config-voiceport)# define tx-bits seize 0101

To configure a voice port on a Cisco MC3810 sending traffic in North American E&M signaling format to convert the signaling to MELCAS format, enter the following commands:

router(config)# voice-port 0/8
router(config-voiceport)# define rx-bits idle 1101
router(config-voiceport)# define rx-bits idle 0101
router(config-voiceport)# define tx-bits seize 1101
router(config-voiceport)# define tx-bits seize 0101

Related Commands

Command
Description

condition

Manipulate the signaling bit-pattern for all voice signaling types.

ignore

Configures an E&M or E&M MELCAS voice port to ignore specific receive bits.


dial-peer hunt

To specify a hunt selection order for dial-peers, use the dial-peer hunt dial-peer configuration command. Use the no form of this command to restore the default selection order.

dial-peer hunt hunt-order-number
no dial-peer hunt

Syntax Description

hunt-order-number

A number from 0 to 7 that selects a predefined hunting selection order:

0—Longest match in phone number, explicit preference, random selection. This is the default hunt order number.

1—Longest match in phone number, explicit preference, least recent use.

2—Explicit preference, longest match in phone number, random selection.

3—Explicit preference, longest match in phone number, least recent use.

4—Least recent use, longest match in phone number, explicit preference.

5—Least recent use, explicit preference, longest match in phone number.

6—Random selection.

7—Least recent use.


Defaults

The default is the longest match in phone number, explicit preference, and random selection (hunt order number 0).

Command Mode

Global configuration

Command History

Release
Modification

12.0(7)XK

This command was introduced, and was first supported on the Cisco 2600 and 3600 series routers and on the Cisco MC3810 multiservice access concentrator.


Usage Guidelines

Use the dial-peer hunt dial-peer configuration command if you have configured hunt groups. "Longest match in phone number" refers to the destination pattern that matches the greatest number of the dialed digits. "Explicit preference" refers to the preference setting in the dial-peer configuration. "Least recent use" refers to the destination pattern that has waited the longest since being selected. "Random selection" weighs all of the destination patterns equally in a random selection mode.

Example

The following example configures the dial peers to hunt in the following order: (1) longest match in phone number, (2) explicit preference, (3) random selection.

router# configure terminal
router(config)# dial-peer hunt 0

Related Commands

Command
Description

preference

Specifies the preferred selection order of a dial peer within a hunt group.

destination-pattern

Specifies the prefix or the complete telephone number for a dial peer.

show dial-peer voice

Displays configuration information for dial peers.


dial-peer terminator

To change the character used as a terminator for variable length dialed numbers, use the dial-peer terminator global configuration command. Use the no form of this command to restore the default terminating character.

dial-peer terminator character
no dial-peer terminator

Syntax Description

character

Designates the terminating character for a variable-length dialed number. Valid numbers and characters are #, *, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, a, b, c, and d. The default is #.


Defaults

The default terminating character is #.

Command Mode

Global configuration

Command History

Release
Modification

12.0

This command was introduced.

12.0(7)XK

Usage was restricted to variable-length dialed numbers.


Usage Guidelines

There are certain areas in the world, 0for example, in certain European countries, where telephone numbers can vary in length. When a dialed-number string is identified as a variable length dialed-number, the system does not place a call until the configured value for the timeouts interdigits command has expired, or until the caller dials the terminating character. Use the dial-peer terminator global configuration command to change the terminating character.

Example

The following example specifies "9" as the terminating character for variable-length dialed numbers:

router# configure terminal
router(config)# dial-peer terminator 9#

Related Commands

Command
Description

answer-address

Specifies the preferred selection order of a dial peer within a hunt group.

destination-pattern

Specifies the prefix or the complete telephone number for a dial peer.

timeouts interdigit

Specifies the interdigit timeout value for a voice port in seconds.

show dial-peer voice

Displays configuration information for dial peers.


dial-peer voice

To enter dial-peer configuration mode and specify the method of voice encapsulation, use the dial-peer voice global configuration command.

For the Cisco 2600 series:

dial-peer voice tag {pots | voip | vofr}
no dial-peer voice tag

For the Cisco 3600 series and the Cisco MC3810:

dial-peer voice tag {pots | voip | vofr | voatm}
no dial-peer voice tag

Syntax Description

tag

A number identifying a particular dial peer. Valid entries are 1 to 2147483647.

pots

POTS dial peer using basic telephone service.

voip

VoIP dial peer using voice encapsulation on the POTS network.

vofr

Voice over Frame Relay dial peer using encapsulation on the Frame Relay backbone network.

voatm

(Cisco 3600 and MC3810 only) Voice over ATM dial peer using real-time AAL5 voice encapsulation on the ATM backbone network.


Defaults

No default behavior or values.

Command Mode

Global configuration

Command History

Release
Modification

11.3(1)T

This command was introduced.

11.3(1)MA

This command was first supported on the Cisco MC3810 with support for POTS, VoFR and VoATM.

12.0(3)XG

This command added VoFR to the Cisco 2600 and 3600 series routers.

12.0(4)T

This command added VoFR to the Cisco 7200 series platform.

12.0(7)XK

This command added VoIP to the Cisco MC3810 and VoATM to the Cisco 3600 series routers. Support for VoHDLC on the Cisco MC3810 was removed in this release.


Usage Guidelines

Use the dial-peer voice global configuration command to switch to the dial-peer configuration mode from the global configuration mode. Use the exit command to exit the dial-peer configuration mode and return to the global configuration mode.

Examples

The following example accesses dial-peer configuration mode and configures a POTS peer identified as dial peer 10:

router# configure terminal
router(config)# dial-peer voice 10 pots

The following example accesses dial-peer configuration mode and configures a VoATM peer identified as dial peer 20:

router# configure terminal
router(config)# dial-peer voice 20 voatm

Related Commands

Command
Description

voice-port

Enters voice-port configuration mode.


disconnect-ack

To configure an FXS voice port to return an acknowledgment upon receipt of a disconnect signal, use the disconnect-ack voice-port configuration command. To disable the acknowledgment, use the no form of this command.

disconnect-ack
no disconnect-ack

Syntax Description

This command has no arguments or keywords.

Defaults

FXS voice ports return an acknowledgment upon receipt of a disconnect signal.

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

This command configures an FXS voice port to remove line power if the equipment on an FXS loop-start trunk disconnects first.

Examples

The following example turns off the disconnect acknowledgment signal on voice port 1/1 on a Cisco MC3810:

router(config)# voice-port 1/1
router(config-voiceport)# no disconnect-ack

The following example turns off the disconnect acknowledgment signal on voice port 1/1/0 on a Cisco 2600 or 3600:

router(config)# voice-port 1/0/0
router(config-voiceport)# no disconnect-ack

Related Commands

Command
Description

show voice port

Displays voice port configuration information.


ds0-group

To specify the DS0 timeslots that make up a logical voice port on a T1 or E1 controller, and to specify the signaling type, use the ds0-group controller configuration command. Use the no form of the command to remove the DS0 group and signaling setting.

ds0-group ds0-group-no timeslots timeslot-list type signal-type

no ds0-group ds0-group-no

Syntax Description

ds0-group-no

A value from 0 to 23 that identifies the DS0 group.

timeslot-list

timeslot-list is a single timeslot number, a single range of numbers, or multiple ranges of numbers separated by commas. For T1, allowable values are from 1 to 24. Examples are:

2

1-15, 17-24

1-23

2, 4, 6-12

type

The signaling method selection for type depends on the connection that you are making. The E&M interface allows connection for PBX trunk lines (tie- lines) and telephone equipment. The FXS interface allows connection of basic telephone equipment and PBXs. The FXO interface is for connecting the central office (CO) to a standard PBX interface where permitted by local regulations. The FXO interface is often used for off-premises extensions.

The options are as follows:

e&m-immediate-start—no specific off-hook and on-hook signaling

e&m-delay-dial—the originating endpoint sends an off-hook signal and then waits for an off-hook signal followed by an on-hook signal from the destination

e&m-wink-start—the originating endpoint sends an off-hook signal and waits for a wink signal from the destination

fxs-ground-start—Foreign Exchange Station ground-start signaling support

fxs-loop-start —Foreign Exchange Station loop-start signaling support

fxo-ground-start—Foreign Exchange Office ground-start signaling support

fxo-loop-start—Foreign Exchange Office loop-start signaling support

 

The following options are available only on E1 controllers on the Cisco MC3810:

e&m-melcas-immed—E&M Mercury Exchange Limited Channel Associated Signaling (MELCAS) immediate start signaling support

e&m-melcas-wink—E&M MELCAS wink start signaling support

e&m-melcas-delay—E&M MELCAS delay start signaling support

fxo-melcas—MELCAS Foreign Exchange Office signaling support

fxs-melcas—MELCAS Foreign Exchange Station signaling support

 

The ext-sig option is available only when the mode ccs command is enabled on the Cisco MC3810 for FRF.11 transparent CCS support.


Defaults

No DS0 group is defined.

Command Mode

Controller configuration

Command History

Release
Modification

11.2

This command was introduced for the Cisco AS5300 as cas-group.

12.0(1)T

The cas-group command was first supported on the Cisco 3600 series.

12.0(5)T

This command was renamed ds0-group on the Cisco AS5300 and on the Cisco 2600 and 3600 series (requires Digital T1 Packet Voice Trunk Network Modules).

12.0(7)XK

Support for this command was extended to the Cisco MC3810. When the ds0-group command became available on the Cisco MC3810, the voice-group command was removed and no longer supported. ext-sig replaces the ext-sig-master and ext-sig-master options that were available with the voice-group command.


Usage Guidelines

The ds0-group command automatically creates a logical voice port that is numbered as follows:

Cisco 2600 and 3600 series:

slot/port:ds0-group-no.

Cisco MC3810:

slot:ds0-group-no

On the Cisco MC3810, the slot number is the controller number. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.

Examples

The following example configures ranges of T1 controller timeslots for FXS ground-start and FXO loop-start signaling on a Cisco 2600 or 3600 Series router:

router(config)# controller T1 1/0
router(config-controller)# framing esf
router(config-controller)# linecode b8zs
router(config-controller)# ds0-group 1 timeslot 1-10 type fxs-ground-start
router(config-controller)# ds0-group 2 timeslot 11-24 type fxo-loop-start

The following example configures DS0 groups 1 and 2 on controller T1 1 on the Cisco MC3810 to support Transparent CCS:

router(config)# controller T1 1
router(config-controller)# mode ccs cross-connect
router(config-controller)# ds0-group 1 timeslot 1-10 type ext-sig
router(config-controller)# ds0-group 2 timeslot 11-24 type ext-sig

Related Commands

Command
Description

codec complexity

Matches the DSP complexity packaging to the codec(s) to be supported

mode ccs

Configures the T1/E1 controller to support CCS cross-connect or CCS frame-forwarding.


encapsulation

To configure the ATM adaptation layer (AAL) and encapsulation type for an ATM PVC class, use the encapsulation command in the appropriate command mode. Use the no form of this command to remove an encapsulation from a PVC class.

encapsulation aal-encap
no encapsulation aal-encap


Note   This document only describes encapsulation settings for Voice over ATM. For the full syntax of the encapsulation command, refer to Cisco IOS 12.0 Wide Area Networking Command Reference.


Syntax Description

aal-encap

ATM adaptation layer (AAL) and encapsulation type. Possible values for aal-encap are as follows:

aal5mux voice—For a MUX-type virtual circuit for Voice over ATM.

aal5snap—The only encapsulation supported for Inverse ARP. Logical Link Control/Subnetwork Access Protocol (LLC/SNAP) precedes the protocol datagram.


Defaults

The global default encapsulation is aal5snap. See the "Usage Guidelines" section for other default characteristics.

Command Mode

Interface-ATM-VC configuration (for an ATM PVC or SVC)

Command History

Release
Modification

11.3 T

This command was introduced.

12.0

This command superseded the encapsulation atm command for the Cisco MC3810, and the aal5mux frame and aal5mux voice suboptions appeared.

12.0(7)XK

Support for the aal5mux voice option was added to the Cisco 3600 series routers.


Usage Guidelines

Use one of the aal5mux encapsulation options to dedicate the specified PVC to a single protocol; use the aal5snap encapsulation option to multiplex two or more protocols over the same PVC. Whether you select aal5mux or aal5snap encapsulation depends on practical considerations, such as the type of network and the pricing offered by the network. If the network's pricing depends on the number of PVCs set up, aal5snap may be the appropriate choice. If pricing depends on the number of bytes transmitted, aal5mux may be the appropriate choice because it has slightly less overhead.

If you specify virtual template parameters after the ATM PVC is configured, issue a shutdown command followed by a no shutdown command on the ATM subinterface to restart the interface, causing the newly configured parameters (such as an IP address) to take effect.

Example

The following example configures a PVC to support encapsulation for Voice over ATM:

router(config-if)# pvc 20
router(config-if-atm-pvc)# encapsulation aal5mux voice

encapsulation ftc-trunk

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

forward-digits

To specify which digits to forward for voice calls, use the forward-digits dial-peer configuration command. If the no form of this command is entered, any digits not matching the destination-pattern are not forwarded. Use the default form of this command to restore the default state.

forward-digits {num-digit | all | extra}
no forward-digits
default forward-digits

Syntax Description

num-digit

The number of digits to be forwarded. If the number of digits is greater than the length of a destination phone number, the length of the destination number is used. The valid range is 0 to 32. Setting the value to 0 is equivalent to entering no forward-digits.

all

Forward all digits. If all is entered, the full length of the destination pattern is used.

extra

If the length of the dialed digit string is greater than the length of the dial-peer destination pattern, the extra right-justified digits are forwarded. However, if the dial-peer destination pattern is variable length (ending with character "T", for example: T, 123T, 123...T), extra digits are not forwarded.


Defaults

Dialed digits not matching the destination-pattern are forwarded.

Command Mode

Dial-peer configuration

Command History

Release
Modification

11.3(1) MA

This command was introduced on the Cisco MC3810.

12.0(2) T

The implicit option was added.

12.0(4) T

This command was modified to support ISDN PRI QSIG signaling calls.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 platforms, the implicit keyword was removed, and the extra keyword was added.


Usage Guidelines

This command applies only to POTS dial peers.

Forwarded digits are always right-justified, so that extra leading digits are stripped.

The destination pattern includes both explicit digits and wildcards if present.

Use the default form of this command if a non-default digit-forwarding scheme was entered previously and you wish to restore the default.

For QSIG ISDN connections, entering forward-digits all implies that all the digits of the called party number are sent to the ISDN connection. When you enter forward-digits num-digit and enter a number from 1 to 32, the number of digits specified (right justified) of the called part number are sent to the ISDN connection.

Examples

The following example forwards all of the digits in the destination pattern of a POTS dial peer:

router(config)# dial-peer voice 1 pots 
router(config-dial-peer)# destination-pattern 8... 
router(config-dial-peer)# forward-digits all

The following example forwards 4 of the digits in the destination pattern of a POTS dial peer:

router(config)# dial-peer voice 1 pots 
router(config-dial-peer)# destination-pattern 555.... 
router(config-dial-peer)# forward-digits 4

The following example forwards the extra right-justified digits that exceed the length of the destination pattern of a POTS dial peer:

router(config)# dial-peer voice 1 pots 
router(config-dial-peer)# destination-pattern 555.... 
router(config-dial-peer)# forward-digits extra

Related Commands

Command
Description

destination-pattern

Defines the prefix or the full E.164 telephone number to be used for a dial peer.

show dial-peer voice

Displays configuration information for dial peers.


frag-pre-queuing

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

frame-relay interface-dlci

To assign a data link connection identifier (DLCI) to a specified Frame Relay subinterface on the router or access server, use the frame-relay interface-dlci interface configuration command. Use the no form of this command to remove this assignment.

frame-relay interface-dlci dlci [ietf | cisco] [voice-cir cir]
no frame-relay interface-dlci dlci [ietf | cisco] [voice-cir cir]

Syntax Description

dlci

DLCI number to be used on the specified subinterface.

ietf | cisco

(Optional) Encapsulation type: Internet Engineering Task Force (IETF) Frame Relay encapsulation or Cisco Frame Relay encapsulation.

voice-cir cir

(Optional; supported on the Cisco MC3810 only.) Specifies the upper limit on the voice bandwidth that may be reserved for this DLCI. The default is the CIR configured for the Frame Relay map class. For more information, see the "Usage Guidelines" section.


Defaults

No DLCI is assigned.

Command Mode

Interface configuration

Command History

Release
Modification

10.0

This command was introduced.

11.3(1) MA

The voice-encap option was added for the Cisco MC3810.

12.0(2) T

The voice-cir option was added for the Cisco MC3810.

12.0(3)XG and 12.0(4)T

Additional usage guidelines added.

12.0(7)XK

The voice-encap option on the Cisco MC3810 was removed. This option is no longer supported.


ftc-trunk frame-relay-dlci

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

ftc-trunk management-dlci

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

ftc-trunk management-protocol

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

ftc-trunk voice-dlci

This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.

huntstop

To disable all further dial-peer hunting if a call fails when using hunt groups, enter the huntstop dial-peer configuration command. To reenable dial-peer call hunting, enter the no form of this command.

huntstop
no huntstop

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled

Command Mode

Dial-peer configuration

Command History

Release
Modification

12.0(5)T

This command was introduced on the Cisco MC3810.

12.0(7)XK

Support for this command was extended to the Cisco 2600 and 3600 series routers.


Usage Guidelines

Once you enter this command, no further hunting is allowed if a call fails on the specified dial peer.

This command can be used with all types of dial peers.

Examples

The following example shows how to disable dial-peer hunting on a specific dial peer:

router(config)# dial peer voice 100 vofr
router(config-dial-peer)# huntstop

The following example shows how to reenable dial-peer hunting on a specific dial peer:

router(config)# dial peer voice 100 vofr
router(config-dial-peer)# no huntstop

Related Commands

Command
Description

dial-peer voice

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.


icpif

To specify the Impairment/Calculated Planning Impairment Factor (ICPIF) for calls sent by a dial peer, use the icpif dial peer configuration command. Use the no form of this command to restore the default value for this command.

icpif number
no icpif number

Syntax Description

number

Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are from 0 to 55.


Defaults

The default value for this command is 30.

Command Mode

Dial-peer configuration

Command History

Release
Modification

11.3(1)T

This command was introduced on the Cisco 3600 series.

12.0(7)XK

This command was first supported on the Cisco MC3810 platform.


Usage Guidelines

Use the icpif command to specify the maximum acceptable impairment factor for the voice calls sent by the selected dial peer.

This command is applicable only to VoIP peers.

Example

The following example disables the icpif command:

dial-peer voice 10 voip
 icpif 0

ignore

To configure the E&M or E&M MELCAS voice port to ignore specific receive bits, use the ignore voice-port configuration command. Use the no form of this command to restore the default value.

ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
no ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}

Syntax Description

rx-a-bit

Ignores the receive A bit.

rx-b-bit

Ignores the receive B bit.

rx-c-bit

Ignores the receive C bit.

rx-d-bit

Ignores the receive D bit.


Defaults

The default is mode-dependent:

E&M:

no ignore rx-a-bit
ignore rx-b-bit, rx-c-bit, rx-d-bit

E&M MELCAS:

no ignore rx-b-bit, rx-c-bit, rx-d-bit

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

This command applies to E&M digital voice ports associated with T1/E1 controllers. Repeat the command for each receive bit to be configured. Use this command with the define command.

Examples

To configure voice-port 1/1 on a Cisco MC3810 to ignore receive bits a, b, and c and to monitor receive bit d, enter the following commands:

router(config)# voice-port 1/1
router(config-voiceport)# ignore rx-a-bit
router(config-voiceport)# ignore rx-b-bit
router(config-voiceport)# ignore rx-c-bit
router(config-voiceport)# no ignore rx-d-bit

To configure voice-port 1/0/0 on a Cisco 3600 to ignore receive bits a, c, and d and to monitor receive bit b, enter the following commands:

router(config)# voice-port 1/0/0
router(config-voiceport)# ignore rx-a-bit
router(config-voiceport)# ignore rx-c-bit
router(config-voiceport)# ignore rx-d-bit
router(config-voiceport)# no ignore rx-b-bit

Related Commands

Command
Description

condition

Manipulates the signaling bit-pattern for all voice signaling types.

define

Defines the transmit and receive bits for E&M and E&M MELCAS voice signaling.

show voice port

Displays configuration information for voice ports.


incoming called-number

To identify the service type for a call on a router handling both voice and modem calls, use the incoming called-number dial peer configuration command. To return to the default value, use the no form of this command.

incoming called-number string
no incoming called-number string

Syntax Description

string

Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number.


Defaults

The default value for this command is no associated called number.

Command Mode

Dial peer configuration

Command History

Release
Modification

11.3NA

This command was introduced on the Cisco AS5800 platform.

12.0(7)XK

This command was first supported on the Cisco MC3810 platform.


Usage Guidelines

When the Cisco MC3810 is handling both modem and voice calls, it needs to be able to identify the service type of the call—meaning whether the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.

If you do not use the incoming called-number command, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.

By default, there is no called number associated with the dial peer, which means that incoming calls will be associated with dial peers based on matching calling number with answer address, call number with destination pattern, or calling interface with configured interface.

This command applies to both VoIP and POTS dial peers.

Example

The following example configures calls coming in to the server with a called number of "3799262" as voice calls:

dial peer voice 10 pots
 incoming called-number 3799262

isdn contiguous-bchan

To configure contiguous bearer channel handling on an E1 Primary Rate Interface (PRI) interface, use the isdn contiguous-bchan interface configuration command. To disable the contiguous B channel handling, use the no form of this command.

isdn contiguous-bchan

no isdn contiguous-bchan

Syntax Description

This command has no arguments or keywords.

Defaults

By default, contiguous B channel handling is off.

Command Mode

Interface configuration

Command History

Release
Modification

12.0(7)XK

This command was introduced.


Usage Guidelines

Use the isdn contiguous-bchan command to specify contiguous bearer channel handling so that B channels 1 through 30, skipping 16, map to timeslots 1 through 31). This is available for E1 PRI interfaces only, when the primary-qsig switch type option is configured by using the isdn switch-type command.

Example

The following example shows the command configuration on a Cisco 3660 series router E1 interface:

interface Serial5/0:15
 no ip address
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn incoming-voice voice
 isdn continguous-bchan

Related Commands

Command
Description

isdn switch-type primary-qsig

In global or interface configuration mode, configures the primary-qsig switch type for PRI support.


isdn incoming-voice

To route all incoming voice calls as voice calls, to route them the modem and treat them as analog data, or to ensure that calls bypass the modems and are treated as digital data, use the isdn incoming-voice interface configuration command. Use the no form of this command to disable the setting.

isdn incoming-voice {data [56 | 64] | modem [56 | 64] | voice}

no isdn incoming-voice {data [56 | 64] | modem [56 | 64] | voice}

Syntax Description

data

Specifies that incoming voice calls bypass the modems and are handled as digital data.

modem

Specifies that incoming voice calls are passed over to the digital modems, where they negotiate the appropriate modem connection with the far-end modem.

voice

Specifies that incoming voice calls are treated as voice calls rather than being routed to the modem or handled as digital data.

56

Specifies that the bandwidth for this connection is 56 kbps.

64

Specifies that the bandwidth for this connection is 64 kbps. If no argument is entered for either the data or modem keywords, the default value is 64.


Defaults

When a PRI or BRI interface is created, isdn incoming-voice voice is the default, except on a Cisco 2600 or 3600 BRI S/T TE interface. In this case, if the command is not specified, the default isdn incoming-voice modem configuration setting is converted to isdn incoming-voice voice when the interface receives an incoming call.

Command Mode

Interface configuration

Command History

Release
Modification

11.1

This command was introduced.

12.0(2)XC and 12.0(3)T

This command was made available for BRI interfaces.

12.0(7)XK

This command was modified to include the voice keyword.


Usage Guidelines

Unless you specify otherwise, all calls received by the router and characterized as voice calls are treated as such and not handled as digital data or not passed over to the modem.

On a Cisco 2600 or 3600 series router BRI S/T TE interface where the isdn incoming-voice command is not specified, the default isdn incoming-voice modem configuration setting is converted to isdn incoming-voice voice when the interface receives an incoming call.

To establish speedier connections for analog calls to the router, use the isdn incoming-voice command with the modem keyword to have voice calls routed through digital modems (as pulse-code modulated analog data) instead of being treated as digital data.

Example

The following example shows the command configuration on a Cisco 3660 series router T1 PRI interface:

interface Serial5/0:23
 no ip address
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn incoming-voice voice

isdn protocol-emulate

To configure a Primary Rate Interface (PRI) interface to serve as either the primary QSIG slave or the primary QSIG master, use the isdn protocol-emulate interface command. To disable QSIG signaling, use the no form of this command.

isdn protocol-emulate { user | network }

no isdn protocol-emulate { user | network }

Syntax Description

user

Enter user (equivalent to the QSIG term slave) to configure the port as the terminating end. This is the default.

network

Enter network (equivalent to the QSIG term master) to configure the port as NT; the PINX is the slave.


Defaults

User

Command Mode

Interface configuration mode.

Command History

Release
Modification

12.0(7)T

This command was introduced for the Cisco AS5300.

12.0(7)XK

This command was introduced for the Cisco MC3810, and for the Cisco 7200 VXR, Cisco 2600, and Cisco 3600 series routers.


Usage Guidelines

On the Cisco MC3810, this command replaces the command isdn switch-type [primary-qsig-slave | primary-qsig-master] command.

Examples

The following example shows the command configuration on a Cisco 3660 series router T1 PRI interface:

interface Serial5/0:23
 no ip address
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn protocol-emulate user

isdn switch type

To specify a central office switch type or configure a Primary Rate Interface (PRI) interface to support Q.SIG signaling, use the isdn switch-type global or interface command. To disable the central office switch type or QSIG signaling, use the no form of this command.

isdn switch-type {switch-type | primary-qsig | basic-qsig}

no isdn switch-type {switch-type | primary-qsig | basic-qsig}

Syntax Description

switch-type

Service provider switch type. See for a list.

primary-qsig

PRI

basic-qsig

BRI


Defaults

The switch type defaults to none, which disables the switch type.

Command Modes

Global configuration mode or interface configuration mode.

Command History

Release
Modification

9.21

Introduced as a global command.

11.3 T

Introduced as an interface command.

12.0(2)T

primary-qsig-slave and primary-qsig-master keywords introduced for the Cisco MC3810.

12.0(7)K

primary-qsig-slave and primary-qsig-master keywords for the Cisco MC3810 are no longer supported. primary-qsig and basic-qsig keywords supported on the Cisco MC3810, Cisco 7200 VXR, 2600 and 3600 series routers.


Usage Guidelines

You can enter the isdn switch-type command to support QSIG at either the global configuration level or at the interface configuration level. For example, if you have a QSIG connection on one line as well as on the BRI or PRI port, you can configure the ISDN switch type in one of the following combinations:

Set the global isdn switch-type command to support QSIG by entering either the isdn-switch-type basic-qsig command (BRI) or isdn-switch-type primary-qsig command (BRI); and set the interface isdn-switch-type command for the interfaces to a regular central office switch type, such as those shown in .

Set the global isdn switch-type command to support the CO switch type (see ), and set the interface isdn switch-type command for the interface to support QSIG.

Configure the global isdn switch-type command to another setting (see ); then, set the interface isdn switch-type command for interface bri to a BRI setting; set the interface isdn switch-type command for the serial interface to support QSIG.

Table 2 ISDN CO Switch Types

Country
ISDN Switch Type
Description

Australia

basic-ts013

Australian TS013 switches

Europe

basic-1tr6

German 1TR6 ISDN switches

 

basic-nwnet3

Norwegian NET3 ISDN switches (phase 1)

 

basic-net3

NET3 ISDN switches (UK and others)

 

vn2

French VN2 ISDN switches

 

vn3

French VN3 ISDN switches

Japan

ntt

Japanese NTT ISDN switches

New Zealand

basic-nznet3

New Zealand NET3 switches

North America

basic-5ess

Lucent Technologies basic rate switches

 

basic-dms100

NT DMS-100 basic rate switches

 

basic-ni1

National ISDN-1 switches


Examples

The following example shows the command configuration on a Cisco 3660 series router T1 PRI interface:

interface Serial5/0:23
 no ip address
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn protocol-emulate user

Related Commands

Command
Description

isdn protocol-emulate

Configures the interface to serve as either the QSIG slave or the QSIG master.


loss-plan

To specify the analog-to-digital gain offset for an analog FXO or FXS voice port, enter the codec dial-peer configuration command. Use the no form of this command to restore the default value.

loss-plan {plan1 | plan2 | plan3 | plan4 | plan5 | plan6 | plan7 | plan8 | plan9}
no loss-plan

Syntax Description

plan1

FXO: A-D gain = 0 dB, D-A gain = 0 dB

FXS: A-D gain = -3 dB, D-A gain = -3 dB

plan2

FXO: A-D gain = 3 dB, D-A gain = 0 dB

FXS: A-D gain = 0 dB, D-A gain = -3 dB

plan3

FXO: A-D gain = -3 dB, D-A gain = 0 dB

FXS: Not applicable

plan4

FXO: A-D gain = -3 dB, D-A gain = -3 dB

FXS: Not applicable

plan5

FXO: Not applicable

FXS: A-D gain = -3 dB, D-A gain = -10 dB

plan6

FXO: Not applicable

FXS: A-D gain = 0 dB, D-A gain = -7 dB

plan7

FXO: A-D gain = 7 dB, D-A gain = 0 dB

FXS: A-D gain = 0 dB, D-A gain = -6 dB

plan8

FXO: A-D gain = 5 dB, D-A gain = -2 dB

FXS: Not applicable

plan9

FXO: A-D gain = 6 dB, D-A gain = 0 dB

FXS: Not applicable


Defaults

FXO: A-D gain = 0 dB, D-A gain = 0 dB (loss plan 1)

FXS: A-D gain = -3 dB, D-A gain = -3 dB (loss plan 1)

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3(1)MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

The following additional signal level choices were added: plan 3, plan 4, plan 8, and plan 9.


Usage Guidelines

This command sets the analog signal level difference (offset) between the analog voice port and the digital signal processor (DSP). Each loss plan specifies a level offset in both directions—from the analog voice port to the DSP (A-D) and from the DSP to the analog voice port (D-A).

Use this command to obtain the required levels of analog voice signals to and from the DSP.

This command is supported only on Cisco MC3810 series concentrators, on FXO and FXS analog voice ports.

Example

The following example configures FXO voice port 1/6 for a -3 dB offset from the voice port to the DSP and a 0 dB offset from the DSP to the voice port:

router(config)# voice-port 1/6 
router(config-voiceport)# loss-plan plan3

The following example configures FXS voice port 1/1 for a 0 dB offset from the voice port to the DSP and a -7 dB offset from the DSP to the voice port:

router(config)# voice-port 1/1 
router(config-voiceport)# loss-plan plan6

Related Commands

Command
Description

impedance

Specifies the terminating impedance of the voice port interface. Used on FXO voice ports in correcting input levels.

input gain

Specifies the gain applied by a voice port to the input signal from the PBX or other customer premises equipment.

output attenuation

Specifies the attenuation applied by a voice port to the output signal toward the PBX or other customer premises equipment.


num-exp

To define a complete telephone number for an extension, use the num-exp global configuration command. Use the no form of this command to cancel a configured number expansion.

num-exp extension-number expanded-number
no num-exp extension-number

Syntax Description

extension-number

expanded-number

Digit(s) defining an extension number to be expanded.

Digit(s) defining the expanded telephone number or destination pattern.


Defaults

No number expansion is configured.

Command Mode

Global configuration

Command History

Release
Modification

11.3(1)T

This command was first introduced on the Cisco 3600 platform.

12.0(3)T

This command was first supported on the Cisco AS5300 platform.

12.0(4)XL

This command was first supported on the Cisco AS5800 platform.

12.0(7)XK

This command was first supported on the Cisco MC3810 platform.


Usage Guidelines

Use the num-exp global configuration command to expand a set of numbers (for example, an extension number) into a destination pattern. With this command, you can map specific extensions and expanded numbers together by explicitly defining each number, or you can define extensions and expanded numbers using variables. You can also use this command to convert seven-digit numbers to numbers containing less than seven digits.

Use a period (.) as a variable or wild card, representing a single number. Use a separate period for each number you want to represent with a wildcard; if you want to replace four numbers in an extension with wildcards, type in four periods.

Example

The following example specifies that extension number 55541 be expanded to 14085555541:

num-exp 55541 14085555541

The following example specifies that all five-digit extensions beginning with 5 be expanded to 1408555 . . . .

num-exp 5.... 1408555....

Related Commands

Command
Description

forward-digits

Specifies which digits to forward for voice calls.

prefix

Specifies a prefix for a dial peer.

dial-peer terminator

Change the character used as a terminator for variable length dialed numbers.


playout delay

To tune the playout buffer to accommodate packet jitter caused by switches in the WAN, use the playout-delay voice-port configuration command. Use the no form of this command to restore the default value.

playout-delay {maximum | nominal} milliseconds

no playout-delay {maximum | nominal}

Syntax Description

maximum

The delay time the DSP allows before starting to discard voice packets. The default is 160 milliseconds.

nominal

The initial (and minimum allowed) delay time the DSP inserts before playing out voice packets. The default is 80 milliseconds

milliseconds

Playout-delay value in milliseconds. The range for maximum playout delay is 40 to 320, and the range for nominal playout delay is 40 to 240.


Defaults

The default maximum delay is 160 milliseconds.
The default nominal delay is 80 milliseconds.

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

If there is excessive break-up of voice due to jitter with the default playout delay settings, increase the delay times. If your network is small and jitter is minimal, decrease the delay times to reduce delay.

Examples

The following example configures a nominal playout delay of 80 milliseconds and a maximum playout delay of 160 milliseconds on voice-port 1/1 on a Cisco MC3810:

router(config)# voice-port 1/1
router(config-voiceport)# playout-delay nominal 80
router(config-voiceport)# playout-delay maximum 160

The following example configures a nominal playout delay of 80 milliseconds and a maximum playout delay of 160 milliseconds on voice-port 1/0/0 on the Cisco 2600 or 3600:

router(config)# voice-port 1/0/0
router(config-voiceport)# playout-delay nominal 80
router(config-voiceport)# playout-delay maximum 160

Related Commands

Command
Description

vad

Enables voice activity detection.


pri-group

To specify a ISDN Primary Rate interface (PRI) on a channelized T1 or E1 controller, enter the pri-group controller configuration command. Enter the no form of this command removes the remove the ISDN-PRI configuration.

pri-group timeslots timeslot-range

no pri-group

Syntax Description

timeslot-range

timeslot-list is a single timeslot number, a single range of values. For T1, the allowable range is from 1 to 23. For E1, the allowable values are from 1 to 15.


Default

There is no ISDN-PRI group configured.

Command Mode

Controller configuration

Command History

Release
Modification

12.0(2)T

The command was introduced for the Cisco MC3810 multiservice access concentrator.

12.0(7)XK

The command was introduced for the Cisco 2600 and 3600 series with a different name and some keyword modifications.


Usage Guidelines

The pri-group command applies to the configuration of Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810 multiservice concentrator and the Cisco 2600 and 3600 series routers.

Before you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1 controller. Only one pri group can be configured on a controller.

Example

The following example configures configures ISDN-PRI on all timeslots of controller E1 1 on a Cisco 2600 series router::

isdn switch-type primary-qsig-master
controller T1 1
pri-group timeslots 1-23

Related Command

Command
Description

isdn switch-type

To configure the Cisco 2600 series router PRI interface to support QSIG signalling, enter this command.


ring cadence

To specify the ring cadence for an FXS voice port, use the ring cadence voice-port configuration command. Use the no form of this command to restore the default value.

ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12] [define pulse interval]}
no ring cadence

Syntax Description

pattern01

2 seconds on, 4 seconds off

pattern02

1 second on, 4 seconds off

pattern03

1.5 seconds on, 3.5 seconds off

pattern04

1 second on, 2 seconds off

pattern05

1 second on, 5 seconds off

pattern06

1 second on, 3 seconds off

pattern07

0.8 second on, 3.2 seconds off

pattern08

1.5 seconds on, 3 seconds off

pattern09

1.2 seconds on, 3.7 seconds off

pattern09

1.2 seconds on, 4.7 seconds off

pattern11

0.4 second on, 0.2 second off, 0.4 second on, 2 seconds off

pattern12

0.4 second on, 0.2 second off, 0.4 second on, 2.6 seconds off

define

User-definable ring cadence pattern. Each number pair specifies one ring-pulse time and one ring-interval time. You must enter numbers in pairs, and you can enter 1 to 6 pairs. The second number in the last pair that you enter specifies the interval between rings.

pulse

A number (1 or 2 digits) specifying ring pulse (on) time in hundreds of milliseconds. The range is 1 to 50, for pulses of 100 ms to 5000 ms.
For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.

interval

A number (1 or 2 digits) specifying ring interval (off) time in hundreds of milliseconds. The range is 1 to 50, for pulses of 100 to 5000 ms.
For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.


Defaults

Ring cadence defaults to the pattern you specify with the cptone command.

Command Mode

Voice-port configuration

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers, and the patternXX syntax was introduced.


Usage Guidelines

The patternXX keyword provides preset ring cadence patterns for use on any platform. The define keyword allows you to create a custom ring cadence. On the Cisco 2600 and 3600 series routers, only one or two pairs of digits can be entered under the define keyword.

Examples

The following example configures the ring cadence for 1 second on and 4 seconds off on voice port 1/1 on a Cisco MC3810:

router(config)# voice-port 1/1 
router(config-voiceport)# ring cadence pattern02

The following example configures the ring cadence for 1 second on, 1 second off, 1 second on, and 5 seconds off on voice port 1/2 on a Cisco MC3810:

voice-port 1/2 
router(config-voiceport)# ring cadence define 10 10 10 50

The following example configures the ring cadence for 1 second on and 2 seconds off on voice port 1/0/0 on a Cisco 2600 or 3600:

router(config)# voice-port 1/0/0 
router(config-voiceport)# ring cadence pattern04

Related Commands

Command
Description

ring frequency

Specifies the ring frequency for an FXS voice port.

cptone

Specifies the default tone, ring, and cadence settings according to country.


session target

To configure a network-specific address for a dial peer, use the session target dial-peer configuration command. Use the no form of this command to disable this feature.

Cisco MC3810 Voice over IP:

session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp | loopback:compressed | loopback:uncompressed}
no session target

Cisco 3600 Voice over ATM:

session target interface pvc {name | vpi/vci | vci}
no session target

Syntax Description

For the Cisco MC3810 Voice over IP:

ipv4:destination-address

IP address of the dial peer.

dns:host-name

Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device.

(Optional) You can use one of the following three wildcards with this keyword when defining the session target for VoIP peers:

$s$.—Indicates that the source destination pattern will be used as part of the domain name.

$d$.—Indicates that the destination number will be used as part of the domain name.

$e$.—Indicates that the digits in the called number will be reversed, periods will be added in-between each digit of the called number, and that this string will be used as part of the domain name.

$u$.—Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name.

loopback:rtp

Indicates that all voice data will be looped back to the originating source. This is applicable for VoIP peers.

loopback:compressed

Indicates that all voice data will be looped back in compressed mode to the originating source. This is applicable for POTS peers.

loopback:uncompressed

Indicates that all voice data will be looped-back in uncompressed mode to the originating source. This is applicable for POTS peers.


For Cisco 3600 series Voice over ATM dial peers:

interface

Interface type and interface number on the router.

pvc

The specific ATM permanent virtual circuit (PVC) for this dial peer.

name

The PVC name.

vpi/vci

ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC.

On the Cisco 3600, if you have the Multiport T1/E1 ATM network module with IMA installed, the valid range for vpi is 0-15, and the valid range for vci is 1-255.

If you have the OC3 ATM Network Module installed, the valid range for vpi is 0-15, and the valid range for vci is 1-1023.

vci

ATM network virtual channel identifier (VCI) of this PVC.


Defaults

Enabled with no IP address or domain name defined.

Command Mode

Dial-peer configuration

Command History

Release
Modification

11.3(1) T

This command was first introduced.

11.3(1) MA

Support was added for VoFR,VoATM and VoHDLC dial peers on the Cisco MC38110.

12.0(3) XG and 12.0(4)T

The cid option was added. Support was added for VoFR dial peers on the Cisco 2600 and Cisco 3600 series routers.

12.0(7)XK

Support was added for VoATM dial peers on the Cisco 3600 series routers. Support was added for VoIP dial peers on the Cisco MC3810. Support for VoHDLC on the Cisco MC3810 was removed in this release.


Usage Guidelines

This command applies to both the Cisco 3600 series and the Cisco MC3810.

Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.

The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origination and the loopback type selected.

The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you need to configure if you have groups of numbers associated with a particular router.

Examples

The following example configures a session target using DNS for a host, "voice_router," in the domain "cisco.com":

dial-peer voice 10 voip
 session target dns:voice_router.cisco.com

The following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number—in this case, the four-digit extension, to identify the dial peer. As in the previous example, the domain is "cisco.com."

dial-peer voice 10 voip
 destination-pattern 1310222....
 session target dns:$u$.cisco.com

The following example configures a session target using dns, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.

dial-peer voice 10 voip
 destination-pattern 13102221111
 session target dns:$d$.cisco.com

The following example configures a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.

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

The following example configures a session target for Voice over ATM on the Cisco 3600 series. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.

router(config)# dial-peer voice 12 voatm
router(config-dial-peer)# destination-pattern 13102221111
router(config-dial-peer)# session target atm1/0 pvc 1/100

Related Commands

Command
Description

called-number

Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.

codec (dial-peer)

Specifies the voice coder rate of speech for a dial peer.

cptone

Specifies a regional tone, ring, and cadence setting for an analog voice port.

destination-pattern

Specifies either the prefix or the full E.164 telephone number to be used for a dial peer.

dtmf-relay

Enables the DSP to generate FRF.11 Annex A frames for a dial peer.

preference

Indicates the preferred selection order of a dial peer within a hunt group.

session protocol

Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.


show call active voice

To show the active call table, use the show call active voice EXEC command.

show call active voice

Syntax Description

This command has no arguments or keywords.

Command Mode

User EXEC

Command History

Release
Modification

11.3(1)T

This command was introduced on the Cisco 2600 and 3600.

12.0(3)XG

Support for VoFR was added.

12.0(4)T

This command was first supported on the Cisco 7200 series.

12.0(7)XK

This command was first supported on the Cisco MC3810 series.


Usage Guidelines

This command applies to Voice over IP, Voice over Frame Relay, and Voice over ATM on the Cisco 2600, 3600, and MC3810 series.

Use this command to display the contents of the active call table, which shows all of the calls currently connected through the router. This command displays information about call times, dial peers, connections, Quality of Service, and other status and statistical information.

See for a listing of the information types associated with this command.

Example

The following is sample output from the show call active voice command:

router# show call active voice
GENERIC: SetupTime=21072 Index=0 PeerAddress= PeerSubAddress= PeerId=0  
PeerIfIndex=0 LogicalIfIndex=0 ConnectTime=0 CallState=3 CallOrigin=2 ChargedUnits=0  
InfoType=0 TransmitPackets=375413 TransmitBytes=7508260 ReceivePackets=377734  
ReceiveBytes=7554680 

VOIP: ConnectionId[0x19BDF910 0xAF500007 0x0 0x58ED0] RemoteIPAddress=17635075  
RemoteUDPPort=16394 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1  
SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=600 
GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=110 
LoWaterPlayoutDelay=64 ReceiveDelay=94 VADEnable=0 CoderTypeRate=0 

GENERIC: SetupTime=21072 Index=1 PeerAddress=+14085271001 PeerSubAddress=  
PeerId=0 PeerIfIndex=0 LogicalIfIndex=5 ConnectTime=21115 CallState=4 CallOrigin=1  
ChargedUnits=0 InfoType=1 TransmitPackets=377915 TransmitBytes=7558300  
ReceivePackets=375594 ReceiveBytes=7511880 

TELE: ConnectionId=[0x19BDF910 0xAF500007 0x0 0x58ED0] TxDuration=16640  
VoiceTxDuration=16640 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=4  
OutSignalLevel=-440 InSignalLevel=-440 InfoActivity=2 ERLLevel=227  
SessionTarget= 

provides an alphabetical listing of the fields in this output and a description of each field.

Table 3 Show Call Active Voice Field Descriptions 

Field
Description

ACOM Level

Current ACOM level for the call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.

CallOrigin

Call origin; answer versus originate.

CallState

Current state of the call.

CoderTypeRate

Negotiated coder transmit rate of voice/fax compression during the call.

ConnectionId

Global call identifier of a gateway call.

ConnectTime

Time at which the call was connected.

Dial-Peer

Tag of the dial peer transmitting this call.

ERLLevel

Current Echo Return Loss (ERL) level for this call.

FaxTxDuration

Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value.

GapFillWithSilence

Duration of voice signal replaced with silence because voice data was lost or not received on time for this call.

GapFillWithPrediction

Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. An example of such pullout is frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms.

GapFillWithInterpolation

Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received on time from voice gateway for this call.

GapFillWithRedundancy

Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received on time from voice gateway for this call.

HiWaterPlayoutDelay

High water mark Voice Playout FIFO Delay during this call.

Index

Dial peer identification number.

InfoActivity

Active information transfer activity state for this call.

InfoType

Information type for this call.

InSignalLevel

Active input signal level from the telephony interface used by this call.

LogicalIfIndex

Index number of the logical interface for this call.

LoWaterPlayoutDelay

Low water mark Voice Playout FIFO Delay during the call.

NoiseLevel

Active noise level for the call.

OnTimeRvPlayout

Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

OutSignalLevel

Active output signal level to telephony interface used by this call.

PeerAddress

Destination pattern associated with this peer.

PeerId

ID value of the peer table entry to which this call was made.

PeerIfIndex

Voice port index number for this peer.

PeerSubaddress

Subaddress to which this call is connected.

ReceiveBytes

Number of bytes received by the peer during this call.

ReceiveDelay

Average Playout FIFO Delay plus the decoder delay during the voice call.

ReceivePackets

Number of packets received by this peer during this call.

RemoteIPAddress

Remote system IP address for the VoIP call.

RemoteUDPPort

Remote system UDP listener port to which voice packets are transmitted.

RoundTripDelay

Voice packet round trip delay between the local and remote system on the IP backbone during the call.

SelectedQoS

Selected RSVP quality of service (QoS) for the call.

SessionProtocol

Session protocol used for an Internet call between the local and remote router via the IP backbone.

SessionTarget

Session target of the peer used for the call.

SetupTime

Value of the System UpTime when the call associated with this entry was started.

TransmitBytes

Number of bytes transmitted from this peer during the call.

TransmitPackets

Number of packets transmitted from this peer during the call.

TxDuration

Duration of transmit path open from this peer to the voice gateway for the call.

VADEnable

Whether or not voice activation detection (VAD) was enabled for this call.

VoiceTxDuration

Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.


Related Commands

Command
Description

show call history voice

Displays the call history table.

show dial-peer voice

Displays configuration information for dial peers.

show num-exp

Displays the number expansions configured.

show voice port

Displays configuration information about a specific voice port.


show call history voice

To display the call history table, use the show call history voice EXEC command.

show call history voice [last number | brief]

Syntax Description

last number

(Optional) Displays the last calls connected, where the number of calls displayed is defined by the argument number. Valid entries for the argument number are numbers from 1 to 2147483647.

brief

(Optional) Displays abbreviated call history information for each leg of a call.


Command Mode

User EXEC

Command History

Release
Modification

11.3(1)T

This command was introduced on the Cisco 3600.

12.0(3)XG

Support for VoFR was added.

12.0(4)T

The brief keyword was added and the command was first supported on the Cisco 7200 series.

12.0(7)XK

Support for brief the keyword was added on the Cisco MC3810 platform.


Usage Guidelines

This command applies to all voice applications on the Cisco 2600, 3600, MC3810, and 7200 platforms.

Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all voice calls connected through this router in descending time order. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number. To display a shortened version of the call history table, use the keyword brief.

Example

The following is sample output from the show call history voice command for a VoFR call using the frf11-trunk session protocol:

router# show call history voice last 1
GENERIC:
SetupTime=8283963 ms
Index=3149
PeerAddress=3623110
PeerSubAddress=
PeerId=3400
PeerIfIndex=18
LogicalIfIndex=0
DisconnectCause=3F
DisconnectText=service or option not available, unspecified
ConnectTime=8283963
DisconectTime=8285463
CallOrigin=1
ChargedUnits=0
InfoType=2
TransmitPackets=94
TransmitBytes=2751
ReceivePackets=0
ReceiveBytes=0
VOFR:
ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852]
Subchannel=[Interface Serial0/0, DLCI 160, CID 10]
SessionProtocol=frf11-trunk
SessionTarget=Serial0/0 160 10
CalledNumber=2603100
VADEnable=ENABLED
CoderTypeRate=g729r8
CodecBytes=30
SignalingType=cas
DTMFRelay=DISABLED
UseVoiceSequenceNumbers=DISABLED
GENERIC:
SetupTime=8283963 ms
Index=3150
PeerAddress=2601100
PeerSubAddress=
PeerId=1100
PeerIfIndex=7
LogicalIfIndex=0
DisconnectCause=3F
DisconnectText=service or option not available, unspecified
ConnectTime=8283964
DisconectTime=8285464
CallOrigin=2
ChargedUnits=0
InfoType=2
TransmitPackets=0
TransmitBytes=-121
ReceivePackets=94
ReceiveBytes=2563
TELE:
ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852]
TxDuration=15000 ms
VoiceTxDuration=2010 ms
FaxTxDuration=0 ms
CoderTypeRate=g729r8
NoiseLevel=-68
ACOMLevel=20
SessionTarget=

The following is sample output from the show call history voice command for a VoIP call:

router# show call history voice
GENERIC: 
SetupTime=20405 
Index=0 
PeerAddress= 
PeerSubAddress= 
PeerId=0 
PeerIfIndex=0 
LogicalIfIndex=0 
DisconnectCause=NORMAL 
DisconnectText= 
ConnectTime=0 
DisconectTime=20595 
CallOrigin=2 
ChargedUnits=0 
InfoType=0 
TransmitPackets=0 
TransmitBytes=0 
ReceivePackets=0 
ReceiveBytes=0

VOIP: 
ConnectionId[0x19BDF910 0xAF500006 0x0 0x56590] 
RemoteIPAddress=17635075 
RemoteUDPPort=16392 
RoundTripDelay=0 
SelectedQoS=0 
SessionProtocol=1 
SessionTarget= 
OnTimeRvPlayout=0 
GapFillWithSilence=0 
GapFillWithPrediction=0
GapFillWithInterpolation=0 
GapFillWithRedundancy=0 
HiWaterPlayoutDelay=0
LoWaterPlayoutDelay=0 
ReceiveDelay=0 
VADEnable=0 
CoderTypeRate=0

TELE: ConnectionId=[0x19BDF910 0xAF500006 0x0 0x56590] 
TxDuration=3030 
VoiceTxDuration=2700 
FaxTxDuration=0 
CoderTypeRate=0 
NoiseLevel=0 
ACOMLevel=0 
SessionTarget=


provides an alphabetical listing of the fields in this output and a description of each field.

Table 4 Show Call History Voice Field Descriptions 

Field
Description

ACOMLevel

Average ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.

CallOrigin

Call origin; answer versus originate.

CoderTypeRate

Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for the call.

ConnectionID

Global call identifier for the gateway call.

ConnectTime

Time the call was connected.

DisconnectCause

Description explaining why the call was disconnected.

DisconnectText

Descriptive text explaining the disconnect reason.

DisconnectTime

Time the call was disconnected.

FaxDuration

Duration of fax transmitted from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing this value by the TxDuration value.

GapFillWithSilence

Duration of voice signal replaced with silence because the voice data was lost or not received on time for this call.

GapFillWithPrediction

Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call.

GapFillWithInterpolation

Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call.

GapFillWithRedundancy

Duration of voice signal played out with signal synthesized from redundancy parameters available because the voice data was lost or not received on time from the voice gateway for this call.

HiWaterPlayoutDelay

High water mark Voice Playout FIFO Delay during the voice call.

Index

Index number identifying the voice-peer for this call.

InfoType

Information type for this call.

LogicalIfIndex

Index of the logical voice port for this call.

LoWaterPlayoutDelay

Low water mark Voice Playout FIFO Delay during the voice call.

NoiseLevel

Average noise level for this call.

OnTimeRvPlayout

Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.

PeerAddress

Destination pattern or number to which this call is connected.

PeerId

ID value of the peer entry table to which this call was made.

PeerIfIndex

Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for the call.

PeerSubAddress

Subaddress to which this call is connected.

ReceiveBytes

Number of bytes received by the peer during this call.

ReceiveDelay

Average Playout FIFO Delay plus the decoder delay during the voice call.

ReceivePackets

Number of packets received by this peer during the call.

RemoteIPAddress

Remote system IP address for the call.

RemoteUDPPort

Remote system UDP listener port to which voice packets for this call are transmitted.

RoundTripDelay

Voice packet round trip delay between the local and remote system on the IP backbone for this call.

SelectedQoS

Selected RSVP quality of service for the call.

SessionProtocol

Session protocol to be used for an Internet call between the local and remote router via the IP backbone.

SessionTarget

Session target of the peer used for the call.

SetUpTime

Value of the System UpTime when the call associated with this entry was started.

TransmitBytes

Number of bytes transmitted by this peer during the call.

TransmitPackets

Number of packets transmitted by this peer during the call.

TxDuration

Duration of the transmit path open from this peer to the voice gateway for the call.

VADEnable

Whether or not voice activation detection (VAD) was enabled for this call.

VoiceTxDuration

Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value.


Related Commands

Command
Description

show call active voice

Displays the contents of the active call table.

show dial-peer voice

Displays configuration information for dial peers.

show num-exp

Displays the number expansions configured.

show voice port

Displays configuration information about a specific voice port.


show num-exp

To show the number expansions configured, use the show num-exp privileged EXEC command.

show num-exp [dialed-number]

Syntax Description

dialed-number

(Optional) Dialed number.


Command Mode

User EXEC and Privileged EXEC

Command History

Release
Modification

11.3(1)T

This command was first introduced on the Cisco 3600 platform.

12.0(3)T

This command was first supported on the Cisco AS5300 platform.

12.0(4)XL

This command was first supported on the Cisco AS5800 platform.

12.0(7)XK

This command was first supported on the Cisco MC3810 platform.


Usage Guidelines

This command applies to VoFR, VoATM, and Voice over IP on the Cisco 2600 series, 3600 series, and MC3810 platforms.

Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.

Example

The following is sample output from the show num-exp command:

router# show num-exp
Dest Digit Pattern = '0...'     Translation = '+14085270...'
Dest Digit Pattern = '1...'     Translation = '+14085271...'
Dest Digit Pattern = '3..'      Translation = '+140852703..'
Dest Digit Pattern = '4..'      Translation = '+140852804..'
Dest Digit Pattern = '5..'      Translation = '+140852805..'
Dest Digit Pattern = '6....'    Translation = '+1408526....'
Dest Digit Pattern = '7....'    Translation = '+1408527....'
Dest Digit Pattern = '8...'     Translation = '+14085288...'

explains the fields in the sample output.

Table 5 Show Num-Exp Voice Field Descriptions

Field
Description

Dest Digit Pattern

Index number identifying the destination telephone number digit pattern.

Translation

Expanded destination telephone number digit pattern.


Related Commands

Command
Description

show call active voice

Displays the contents of the active call table.

show call history voice

Displays the call history table.

show dial-peer voice

Displays configuration information for dial peers.

show voice port

Displays configuration information about a specific voice port.


show voice call

To show the call status for voice ports on the Cisco router or concentrator, use the show voice call EXEC command.

For the Cisco 2600 and 3600 series with analog voice ports:

show voice call [slot/subunit/port | summary]

For the Cisco 2600 and 3600 series with digital voice ports (with T1 packet voice trunk network modules):

show voice call [slot/port:ds0-group | summary]

For the Cisco MC3810 series with analog voice ports:

show voice call [slot/port | summary]

For the Cisco MC3810 series with digital voice ports:

show voice call [slot:ds0-group | summary]

Syntax Description

summary

(Optional) Show a summary of the call status, not the detailed report.

voice-port

(Optional) Displays the call status for a specified voice port.


Command Mode

User EXEC

Command History

Release
Modification

11.3 MA

This command was introduced for the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600 series routers.


Usage Guidelines

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over IP.

This command shows call-processing and protocol state-machine information for a voice port, if it is available. It also shows information on the DSP channel associated with the voice port, if it is available. All real-time information in the DSP channel, such as jitter and buffer overrun for example, is queried to the DSP channel, and asynchronous responses are returned to the host side.

If no call is active on a voice port, the show voice call summary command displays only the VPM (shutdown) state. If a call is active on a voice port, the VTSPS state is shown. For an on-net call or a local call without local-bypass (not cross-connected), the CODEC and VAD fields are displayed. For an off-net call or a local call with local-bypass, the CODEC and VAD fields are not displayed.

CODEC and VAD are not displayed in the show voice call port command, because this information is in the summary display.

This command provides the status at these levels of the call handling module:

Call processing state machine

Protocol state machine

Sample Display

The following is a sample display from the show voice call summary command for voice ports on a Cisco MC3810, showing two local calls connected without local bypass:

router# show voice call summary
PORT    CODEC    VAD VTSP STATE            VPM STATE
======= ======== === ===================== ========================
0:17.18                                     *shutdown*
0:18.19 g729ar8   n  S_CONNECT             FXOLS_OFFHOOK
0:19.20                                    FXOLS_ONHOOK
0:20.21                                    FXOLS_ONHOOK
0:21.22                                    FXOLS_ONHOOK
0:22.23                                    FXOLS_ONHOOK
0:23.24                                    EM_ONHOOK
1/1                                        FXSLS_ONHOOK
1/2                                        FXSLS_ONHOOK
1/3                                        EM_ONHOOK
1/4                                        EM_ONHOOK
1/5                                        FXOLS_ONHOOK
	1/6     g729ar8   n  S_CONNECT             FXOLS_CONNECT

The following is a sample display from the show voice call summary command for voice ports on a Cisco MC3810, showing two local calls connected with local bypass:

router# show voice call summary
PORT    CODEC    VAD VTSP STATE            VPM STATE
======= ======== === ===================== ========================
0:17.18                                     *shutdown*
0:18.19              S_CONNECT             FXOLS_OFFHOOK
0:19.20                                    FXOLS_ONHOOK
0:20.21                                    FXOLS_ONHOOK
0:21.22                                    FXOLS_ONHOOK
0:22.23                                    FXOLS_ONHOOK
0:23.24                                    EM_ONHOOK
1/1                                        FXSLS_ONHOOK
1/2                                        FXSLS_ONHOOK
1/3                                        EM_ONHOOK
1/4                                        EM_ONHOOK
1/5                                        FXOLS_ONHOOK
1/6                  S_CONNECT             FXOLS_CONNECT

The following is a sample display from the show voice call command for analog voice ports on a Cisco MC3810:

router# show voice call
1/1 vpm level 1 state = FXSLS_ONHOOK
vpm level 0 state = S_UP
1/2 vpm level 1 state = FXSLS_ONHOOK
vpm level 0 state = S_UP
1/3 is shutdown
1/4 vtsp level 0 state = S_CONNECT
vpm level 1 state = S_TRUNKED
vpm level 0 state = S_UP
1/5 vpm level 1 state = EM_ONHOOK
vpm level 0 state = S_UP
1/6 vpm level 1 state = EM_ONHOOK
vpm level 0 state = S_UP
sys252#show voice call 1/4
1/4 vtsp level 0 state = S_CONNECT
vpm level 1 state = S_TRUNKED
vpm level 0 state = S_UP
router#			***DSP VOICE VP_DELAY STATISTICS***
Clk Offset(ms): 1445779863, Rx Delay Est(ms): 95
Rx Delay Lo Water Mark(ms): 95, Rx Delay Hi Water Mark(ms): 125
		***DSP VOICE VP_ERROR STATISTICS***
Predict Conceal(ms): 10, Interpolate Conceal(ms): 0
Silence Conceal(ms): 0, Retroact Mem Update(ms): 0
Buf Overflow Discard(ms): 20, Talkspurt Endpoint Detect Err: 0
		***DSP VOICE RX STATISTICS***
Rx Vox/Fax Pkts: 537, Rx Signal Pkts: 0, Rx Comfort Pkts: 0
Rx Dur(ms): 50304730, Rx Vox Dur(ms): 16090, Rx Fax Dur(ms): 0
Rx Non-seq Pkts: 0, Rx Bad Hdr Pkts: 0
Rx Early Pkts: 0, Rx Late Pkts: 0
		***DSP VOICE TX STATISTICS***
Tx Vox/Fax Pkts: 567, Tx Sig Pkts: 0, Tx Comfort Pkts: 0
Tx Dur(ms): 50304730, Tx Vox Dur(ms): 17010, Tx Fax Dur(ms): 0
		***DSP VOICE ERROR STATISTICS***
Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0
		***DSP LEVELS***
TDM Bus Levels(dBm0): Rx -70.3 from PBX/Phone, Tx -68.0 to PBX/Phone
TDM ACOM Levels(dBm0): +2.0, TDM ERL Level(dBm0): +5.6
TDM Bgd Levels(dBm0): -71.4, with activity being voice

Related Commands

Command
Description

show dial-peer voice

Displays the configuration for all VoIP and POTS dial peers configured on the router.

show voice dsp

Shows the current status of all DSP voice channels.

show voice port

Displays configuration information about a specific voice port.


show voice dsp

To show the configuration status for all configured DSP voice channels on the Cisco router or concentrator, use the show voice dsp EXEC command.

show voice dsp

Syntax Description

This command has no arguments or keywords.

Command Mode

User EXEC

Command History

Release
Modification

11.3 MA

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was first supported on the Cisco 2600 and 3600, and the display format was modified.


Usage Guidelines

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over IP.

Use this command when abnormal behavior in the DSP voice channels occurs.

Sample Display

The following is a sample display from the show voice dsp command on a Cisco MC3810:

Router#show voice dsp
                                BOOT                      PAK
TYPE DSP CH CODEC    VERS STATE STATE   RST AI PORT    TS ABORT   TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 001 01 {high}    3.3 idle  idle      6  0                0       1365/1364
         02 {high}        idle                                0             0/0
C549 002 01 {high}    3.3 idle  idle      6  0                0       1365/1364
         02 {high}        idle                                0             0/0
C549 003 01 {high}    3.3 idle  idle      6  0                0       1365/1364
         02 {high}        idle                                0             0/0
C549 004 01 {high}    3.3 idle  idle      6  0                0       1365/1364
         02 {high}        idle                                0             0/0
C549 005 01 {high}    3.3 idle  idle      6  0                0       1365/1364
         02 {high}        idle                                0             0/0
C549 006 01 {high}    3.3 idle  idle      6  0                0       1365/1364
         02 {high}        idle                                0             0/0


provides an alphabetical listing of the fields in this output and a description of each field.

Table 6 Show Voice DSP Field Descriptions 

Field
Description

AI

Number of alarm indications received from the DSP, which may point to abnormality of DSP firmware.

BOOT STATE

Applicable to Cisco MC3810 only of dynamic reload of DSP is permitted.

CH

Voice channel number in DSP.

CODEC

Cisco MC3810 with HCM and Cisco 2600 and 3600 digital:

If the DSP channel is in use, this indicates what codec it is using. If a DSP channel is not in use, this indicates the complexity level configured.

Cisco MC3810 with VCM and Cisco 2600 and 3600 analog:

Indicates what codec is loaded.

DSP

DSP number.

PAK ABORT

The number of DSP packets dropped due to DSP failure in picking up packets from the host.

PORT

The port number associated with the DSP channel. This is a fixed port number on the Cisco 2600 and 3600; this number may change with each new call on the Cisco MC3810.

RST

The number of DSP resets since the most recent clear counters entry.

STATE

The busy/idle state of the DSP channel.

TS

The backplane timeslot associated with this DSP channel. This is a fixed timeslot on the Cisco 2600 and 3600; this number may change with each new call on the Cisco MC3810.

TX/RX-PAK-CNT

An ordered pair of transmit and receive packet counts processed by the DSP since the previous clear counters command was entered.

TYPE

DSP hardware type.

VERS

Version and revision of DSP hardware, in X,Y format.


Related Commands

Command
Description

clear counters

Clears all the current interface counters from the interface.

show voice port

Displays configuration information about a specific voice port.


show voice port

To display configuration information about a specific voice port, use the show voice port EXEC command.

For the Cisco 2600 and 3600 series with analog voice ports:

show voice port [slot/subunit/port | summary]

For the Cisco 2600 and 3600 series with digital voice ports (with T1 packet voice trunk network modules):

show voice port [slot/port:ds0-group | summary]

For the Cisco MC3810 series with analog voice ports:

show voice port [slot/port | summary]

For the Cisco MC3810 series with digital voice ports:

show voice port [slot:ds0-group | summary]

Syntax Description

For the Cisco 2600 and 3600 series with analog voice ports:

slot/subunit/port

(Optional) Displays information for the analog voice port you specify with the slot/subunit/port designation.

slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform.

subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.)

port specifies an analog voice port number. Valid entries are 0 and 1.

summary

(Optional) Displays a summary of all voice ports.


For the Cisco 2600 and 3600 series with digital voice ports:

slot/port:ds0-group

(Optional) Displays information for the digital voice port you specify with the slot/port:ds0-group designation.

slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform.

port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. (One VWIC fits in an NM.)

ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.

summary

(Optional) Displays a summary of all voice ports.


For the Cisco MC3810 series with analog voice ports:

slot/port

(Optional) Displays information for the analog voice port you specify with the slot/port designation.

slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810.

port specifies an analog voice port number. Valid entries are 1 to 6.

summary

(Optional) Displays a summary of all voice ports.


For the Cisco MC3810 series with digital voice ports:

slot:ds0-group

(Optional) Displays information for the digital voice port you specify with the slot:ds0-group designation.

slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1).

ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1.

summary

(Optional) Displays a summary of all voice ports.


Command Mode

User EXEC

Command History

Release
Modification

11.3(1) T

This command was introduced.

12.0(5)XK and 12.0(7)T

The ds0-group argument was added for the Cisco 2600 and 3600 series routers.

12.0(7)XK

The summary keyword was added for the Cisco 2600 and 3600 series routers. The ds0-group argument was added for the Cisco MC3810.


Usage Guidelines

Use the show voice port privileged EXEC command to display configuration and voice-interface-card-specific information about a specific port.

Example

The following is sample output from the show voice port summary command for all voice ports on a Cisco MC3810 with an analog voice module (AVM):

router# show voice port summary
IN  OUT   ECHO
PORT SIG-TYPE     ADMIN OPER IN-STATUS OUT-STATUS GAIN ATTN CANCEL
1/1  fxs-ls       up    up   on-hook    idle        0    0    y
1/2  fxs-ls       up    up   on-hook    idle        0    0    y
1/3  e&m-wnk      up    up   idle       idle        0    0    y
1/4  e&m-wnk      up    up   idle       idle        0    0    y
1/5  fxo-ls       up    up   idle       on-hook     0    0    y
1/6  fxo-ls       up    up   idle       on-hook     0    0    y

The following is sample output from the show voice port summary command on a Cisco MC3810 with a digital voice module (DVM):

                                  IN      OUT
PORT   CH SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
====== == ========== ===== ==== ======== ======== ==
0:17   18 fxo-ls     down  down idle     on-hook  y
0:18   19 fxo-ls     up    dorm idle     on-hook  y
0:19   20 fxo-ls     up    dorm idle     on-hook  y
0:20   21 fxo-ls     up    dorm idle     on-hook  y
0:21   22 fxo-ls     up    dorm idle     on-hook  y
0:22   23 fxo-ls     up    dorm idle     on-hook  y
0:23   24 e&m-imd    up    dorm idle     idle     y
1/1    -- fxs-ls     up    dorm on-hook  idle     y
1/2    -- fxs-ls     up    dorm on-hook  idle     y
1/3    -- e&m-imd    up    dorm idle     idle     y
1/4    -- e&m-imd    up    dorm idle     idle     y
1/5    -- fxo-ls     up    dorm idle     on-hook  y
1/6    -- fxo-ls     up    dorm idle     on-hook  y
Elements :
sys/voip/ccvpm		                vpm_htsp.c (107) 
sys/voip/ccvtsp			               vtsp_core.c (167) 
sys/voip/cli			                  voiceport_action.c (58)

The following is sample output from the show voice port command for an E&M analog voice port on a Cisco 3600:

router# show voice port 1/0/0
E&M Slot is 1, Sub-unit is 0, Port is 0
 Type of VoicePort is E&M
 Operation State is unknown
 Administrative State is unknown
 The Interface Down Failure Cause is 0
 Alias is NULL
 Noise Regeneration is disabled
 Non Linear Processing is disabled
 Music On Hold Threshold is Set to 0 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is disabled
 Echo Cancel Coverage is set to 16ms
 Connection Mode is Normal
 Connection Number is 
 Initial Time Out is set to 0 s
 Interdigit Time Out is set to 0 s
 Analog Info Follows:
 Region Tone is set for northamerica
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 
 Voice card specific Info Follows:
 Signal Type is wink-start
 Operation Type is 2-wire
 Impedance is set to 600r Ohm
 E&M Type is unknown
 Dial Type is dtmf
 In Seizure is inactive
 Out Seizure is inactive
 Digit Duration Timing is set to 0 ms
 InterDigit Duration Timing is set to 0 ms
 Pulse Rate Timing is set to 0 pulses/second
 InterDigit Pulse Duration Timing is set to 0 ms
 Clear Wait Duration Timing is set to 0 ms
 Wink Wait Duration Timing is set to 0 ms
 Wink Duration Timing is set to 0 ms
 Delay Start Timing is set to 0 ms
 Delay Duration Timing is set to 0 ms

The following is sample output from the show voice port command for an FXS analog voice port on a Cisco 3600:

router# show voice port 1/0/0
Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0
 Type of VoicePort is FXS
 Operation State is DORMANT
 Administrative State is UP
 The Interface Down Failure Cause is 0
 Alias is NULL
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 Music On Hold Threshold is Set to 0 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancel Coverage is set to 16ms
 Connection Mode is Normal
 Connection Number is
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
Analog Info Follows:
 Region Tone is set for northamerica
 Currently processing none
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 Voice card specific Info Follows:
 Signal Type is loopStart
 Ring Frequency is 25 Hz
 Hook Status is On Hook
 Ring Active Status is inactive
 Ring Ground Status is inactive
 Tip Ground Status is inactive
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Hook Flash Duration Timing is set to 600 ms 

The following is sample output from the show voice port command for an FXS analog voice port on a Cisco MC3810:

router# show voice port 1/2
Voice port 1/2 Slot is 1, Port is 2
 Type of VoicePort is FXS
 Operation State is UP
 Administrative State is UP
 No Interface Down Failure
 Description is not set
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancel Coverage is set to 8 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Coder Type is g729ar8
 Companding Type is u-law
 Voice Activity Detection is disabled
 Ringing Time Out is 180 s
 Wait Release Time Out is 30 s
 Nominal Playout Delay is 80 milliseconds
 Maximum Playout Delay is 160 milliseconds
 Analog Info Follows:
 Region Tone is set for northamerica
 Currently processing Voice
 Maintenance Mode Set to None (not in mtc mode)
 Number of signaling protocol errors are 0
 Impedance is set to 600r Ohm
 Analog interface A-D gain offset = -3 dB 
 Analog interface D-A gain offset = -3 dB 
 Voice card specific Info Follows:
 Signal Type is loopStart
 Ring Frequency is 20 Hz
 Hook Status is On Hook
 Ring Active Status is inactive
 Ring Ground Status is inactive
 Tip Ground Status is active
 Digit Duration Timing is set to 100 ms
 InterDigit Duration Timing is set to 100 ms
 Ring Cadence are [20 40] * 100 msec
 InterDigit Pulse Duration Timing is set to 500 ms

The following is sample output from the show voice port command for an E&M digital voice port on a Cisco 3600:

router# show voice port 1/0:1
receEive and transMit Slot is 1, Sub-unit is 0, Port is 1
 Type of VoicePort is E&M
 Operation State is DORMANT
 Administrative State is UP
 No Interface Down Failure
 Description is not set
 Noise Regeneration is enabled
 Non Linear Processing is enabled
 Music On Hold Threshold is Set to -38 dBm
 In Gain is Set to 0 dB
 Out Attenuation is Set to 0 dB
 Echo Cancellation is enabled
 Echo Cancel Coverage is set to 8 ms
 Connection Mode is normal
 Connection Number is not set
 Initial Time Out is set to 10 s
 Interdigit Time Out is set to 10 s
 Region Tone is set for US

explains the fields in the sample output.

Table 7 Show Voice Port Field Descriptions 

Field
Description

Administrative State

Administrative state of the voice port.

Alias

User-supplied alias for this voice port.

Analog interface A-D gain offset

Offset of the gain for analog-to-digital conversion.

Analog interface D-A gain offset

Offset of the gain for digital-to-analog conversion.

Clear Wait Duration Timing

Time of inactive seizure signal to declare call cleared.

Coder Type

Voice compression mode used.

Companding Type

Companding standard used to convert between analog and digital signals in PCM systems.

Connection Mode

Connection mode of the interface.

Connection Number

Full E.164 telephone number used to establish a connection with the trunk or PLAR mode.

Currently Processing

Type of call currently being processed: none, voice, or fax.

Delay Duration Timing

Maximum delay signal duration for delay dial signaling.

Delay Start Timing

Timing of generation of delayed start signal from detection of incoming seizure.

Description

Description of the voice port.

Dial Type

Out-dialing type of the voice port.

Digit Duration Timing

DTMF Digit duration in milliseconds.

E&M Type

Type of E&M interface.

Echo Cancel Coverage

Echo Cancel Coverage for this port.

Echo Cancellation

Whether or not echo cancellation is enabled for this port.

Hook Flash Duration Timing

Maximum length of hook flash signal.

Hook Status

Hook status of the FXO/FXS interface.

Impedance

Configured terminating impedance for the E&M interface.

In Gain

Amount of gain inserted at the receiver side of the interface.

In Seizure

Incoming seizure state of the E&M interface.

Initial Time Out

Amount of time the system waits for an initial input digit from the caller.

InterDigit Duration Timing

DTMF interdigit duration in milliseconds.

InterDigit Pulse Duration Timing

Pulse dialing interdigit timing in milliseconds.

Interdigit Time Out

Amount of time the system waits for a subsequent input digit from the caller.

Maintenance Mode

Maintenance mode of the voice port.

Maximum Playout Delay

The amount of time before the Cisco MC3810 DSP starts to discard voice packets from the DSP buffer.

Music On Hold Threshold

Configured Music-On-Hold Threshold value for this interface.

Noise Regeneration

Whether or not background noise should be played to fill silent gaps if VAD is activated.

Nominal Playout Delay

The amount of time the Cisco MC3810 DSP waits before starting to play out the voice packets from the DSP buffer.

Non-Linear Processing

Whether or not non-linear processing is enabled for this port.

Number of signaling protocol errors

Number of signaling protocol errors.

Operations State

Operation state of the port.

Operation Type

Operation of the E&M signal: two-wire or four-wire.

Out Attenuation

Amount of attenuation inserted at the transmit side of the interface.

Out Seizure

Outgoing seizure state of the E&M interface.

Port

Port number for this interface associated with the voice interface card.

Pulse Rate Timing

Pulse dialing rate in pulses per second (pps).

Region Tone

Configured regional tone for this interface.

Ring Active Status

Ring active indication.

Ring Cadence

Configured ring cadence for this interface.

Ring Frequency

Configured ring frequency for this interface.

Ring Ground Status

Ring ground indication.

Ringing Time Out

Ringing time out duration.

Signal Type

Type of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial.

Slot

Slot used in the voice interface card for this port.

Sub-unit

Subunit used in the voice interface card for this port.

Tip Ground Status

Tip ground indication.

Type of VoicePort

Type of voice port: FXO, FXS, and E&M.

The Interface Down Failure Cause

Text string describing why the interface is down,

Voice Activity Detection

Whether Voice Activity Detection is enabled or disabled.

Wait Release Time Out

The time a voice port stays in the call-failure state while the Cisco MC3810 sends a busy tone, reorder tone, or an out-of-service tone to the port.

Wink Duration Timing

Maximum wink duration for wink start signaling.

Wink Wait Duration Timing

Maximum wink wait duration for wink start signaling.


Related Commands

Command
Description

show voice call

Displays the call status for all voice ports on the Cisco router or concentrator.

show call history voice

Displays the call history table.

show dial-peer voice

Displays configuration information about dial peers.

show num-exp

Displays the number expansions that are configured.


show voice trunk-conditioning signaling

To display the status of trunk-conditioning signaling and timing parameters for a voice port, use the show voice trunk-conditioning signaling EXEC command.

show voice trunk-conditioning signaling [summary | voice-port]

Syntax Description

summary

(Optional) Show a summary of the status for all voice ports on the router or concentrator.

voice-port

(Optional) Show a detailed report for a specified voice port.


Command Mode

EXEC

Command History

Release
Modification

12.0(3)XG and 12.0(4)T

This command was introduced on the Cisco MC3810 as show voice permanent-call.

12.0(7)XK

This command was renamed show voice trunk-conditioning signaling.


Usage Guidelines

This command displays the trunk signaling status for analog and digital voice ports on Cisco MC3810 concentrators.

Sample Display

The following is a sample display from the show voice trunk-conditioning signaling summary command for voice ports on a Cisco MC3810:

router# show voice trunk-conditioning signaling summary
1/1 is shutdown
1/4 is shutdown
1/5 : 
TX INFO :slow-mode seq#= 25, sig pkt cnt= 40, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS signal path  is OPEN
RX INFO :slow-mode, sig pkt cnt= 36, prev-seq#= 25, last-ABCD=0000

The following is a sample display from the show voice trunk-conditioning signaling command for voice port 1/5 on a Cisco MC3810:

router# show voice trunk-conditioning signaling 1/5
1/5 : 
TX INFO :slow-mode seq#= 25, sig pkt cnt= 42, last-ABCD=0000
hardware-state ACTIVE signal type is NorthamericanCAS
signal path  is OPEN
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
RX INFO :slow-mode, sig pkt cnt= 37
missing = 0, out of seq = 0, very late = 0 
playout depth = 0 (ms), refill count = 1
prev-seq#= 25, last-ABCD=0000
trunk_down_timer = 4212 (ms), idle timer = 0 (sec),
tx_oos_timer = 0 (sec), rx_ais_duration = 0 (ms)
forced playout signal pattern = NONE
signaling playout history
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
0000 0000 0000 0000 0000 0000 0000 0000 0000 0000

The following is a sample display from the show voice trunk-conditioning signaling summary command for voice ports on a Cisco 3600:

router# show voice trunk-conditioning signaling summary
2/0/0 is shutdown
2/0/1 is shutdown
3/0:0 8  is shutdown
3/0:1 1  is shutdown
3/0:2 2  is shutdown
3/0:3 3  is shutdown
3/0:5 5  is shutdown
3/0:6(6) : 
 status :
3/0:7 7  is shutdown
3/1:0 8  is shutdown
3/1:1 1  is shutdown
3/1:3 3  is shutdown
3/1:5 5  is shutdown
3/1:7 7  is shutdown

The following is a sample display from the show voice trunk-conditioning signaling command for voice port 3/0:6 on a Cisco 3600:

router# show voice trunk-conditioning signaling 3/0:6
hardware-state ACTIVE signal type is NorthamericanCAS
status :
forced playout pattern = STOPPED
trunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0


explains the fields in the sample output.

Table 8 Field Descriptions for show voice trunk-conditioning signaling Command 

Field
Description

current timer

Time since last signaling packets were received.

forced playout pattern

Which forced playout pattern is sent to PBX:

0 = no forced playout pattern is sent

1 = receive IDLE playout pattern is sent

2 = receive OOS playout pattern is sent

hardware-state

Hardware state based on received IDLE pattern:

IDLE = both sides are idle

ACTIVE = at least one side is active

signal type

Signaling type used by lower level driver: Northamerica, MELCAS, transparent, or external.

idle timer

Time the hardware on both sides has been in idle state.

last-ABCD

Last received or transmitted signal bit pattern.

max inter-arrival time

Maximum interval between received signaling packets.

missing

Number of missed signal packets.

mode

Signaling packet generation frequency:

fast mode = every 4 milliseconds

slow mode = same frequency as keepalive timer

out of seq

Number of out-of-sequence signal packets.

playout depth

Number of packets in playout buffer.

prev-seq#

Sequence number of previous signaling packet.

refill count

Number of packets created to maintain nominal length of playout packet buffer.

rx_ais_duration

Time since receipt of AIS indicator.

seq#

Sequence number of signaling packet.

sig pkt cnt

Number of transmitted or received signaling packets.

signal path

Status of signaling path.

signaling playout history

Signaling bits received in last 60 milliseconds.

trunk_down_timer

Time since last signaling packets were received.

tx_oos_timer

Time since PBX started sending OOS signaling pattern defined by signal pattern oos transmit.

very late

Number of very late signaling packets.


Related Commands

Command
Description

show dial-peer voice

Displays the configuration for all VoIP and POTS dial peers configured on the router.

show voice dsp

Shows the current status of all DSP voice channels.

show voice port

Displays configuration information about a specific voice port.

show voice trunk-conditioning supervisory

Displays the status of trunk supervision and configuration parameters for voice ports.


show voice trunk-conditioning supervisory

To display the status of trunk supervision and configuration parameters for voice ports, use the show voice trunk-conditioning supervisory EXEC command.

show voice trunk-conditioning supervisory [summary | voice-port]

Syntax Description

summary

(Optional) Show a summary of the status for all voice ports on the router or concentrator.

voice-port

(Optional) Show a detailed report for a specified voice port.


Command Mode

EXEC

Command History

Release
Modification

12.0(7)XK

This command was introduced on the Cisco MC3810.


Usage Guidelines

This command displays the trunk supervision and configuration status for analog and digital voice ports.

Sample Display

The following is a sample display from the show voice trunk-conditioning supervisory summary command for voice ports on a Cisco MC3810:

router# show voice trunk-conditioning supervisory summary
1/1 is shutdown
1/4 is shutdown
1/5 : state : TRUNK_SC_CONNECT, voice : on , signal : on ,slave

The following is a sample display from the show voice trunk-conditioning supervisory command for voice port 1/5 on a Cisco MC3810:

router# show voice trunk-conditioning supervisory 1/5
1/5 : state : TRUNK_SC_CONNECT, voice : on, signal : on, slave 
	status: trunk connected
	sequence oos : idle and oos
	pattern :rx_idle = 0x0 rx_oos = 0xF tx_oos =  0xF
	timing : idle = 0, restart = 0, standby = 0, timeout = 40
	supp_all = 50, supp_voice = 0, keep_alive = 5
	timer: oos_ais_timer = 0, timer = 0

The following is a sample display from the show voice trunk-conditioning supervisory summary command for voice ports on a Cisco 3600:

router# show voice trunk-conditioning supervisory summary
2/0/0 is shutdown
2/0/1 is shutdown
3/0:0 8  is shutdown
3/0:1 1  is shutdown
3/0:2 2  is shutdown
3/0:3 3  is shutdown
3/0:5 5  is shutdown
3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/0:7(7) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:0(8) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:1(1) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:3(3) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master
3/1:5(5) is shutdown
3/1:7(7) is shutdown

The following is a sample display from the show voice trunk-conditioning supervisory command for voice port 3/0:6 on a Cisco 3600:

router# show voice trunk-conditioning supervisory 3/0:6
3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on, signal : on, master
	status: trunk connected
	sequence oos : idle and oos
	pattern :rx_idle = 0x0 rx_oos = 0xF 
	timing : idle = 0, restart = 0, standby = 0, timeout = 40
	supp_all = 0, supp_voice = 0, keep_alive = 5
	timer: oos_ais_timer = 0, timer = 0

explains the fields in the sample output.

Table 9 Field Descriptions for show voice trunk-conditioning supervisory Command 

Field
Description

keep_alive

Signaling packets periodically sent to the far end, even if there is no signal change. These signaling packets function as keepalive messages.

master

The voice port configured as connection trunk xxxx.

slave

The voice port configured as connection trunk xxxx answer-mode.

oos_ais_timer

Time since the signaling packet with AIS indicator was received.

pattern

4-bit signaling pattern.

restart

The restart timeout after far end is OOS.

rx-idle

The signaling bit pattern indicating that the far end is idle.

rx-oos

The signaling bit pattern sent to the PBX indicating that the network is OOS.

standby

The time before the slave side goes back to standby after far end goes OOS.

supp_all

The timeout before suppressing transmission of voice and signaling packets to the far end after detection of PBX OOS.

supp_voice

The timeout before suppressing transmission of voice packet to the far end after detection of PBX OOS.

timeout

The timeout for non-receipt of keepalive packets before the far end is considered to be OOS.

TRUNK_SC_CONNECT

Trunk conditioning supervisory component status.


Related Commands

Command
Description

show dial-peer voice

Displays the configuration for all dial peers configured on the router.

show voice dsp

Shows the current status of all DSP voice channels.

show voice port

Displays configuration information about a specific voice port.

show voice trunk-conditioning signaling

Displays the status of trunk-conditioning signaling and timing parameters for a voice port.


signal pattern

To configure the ABCD signaling bit pattern for Cisco trunks and FRF.11 trunks, use the signal pattern voice-class configuration command. Use the no form of this command to restore the default.

signal pattern {idle receive | idle transmit | oos receive | oos transmit} bit-pattern
no signal pattern {idle receive | idle transmit | oos receive | oos transmit}

Syntax Description

idle receive

Defines the signaling pattern for identifying an "idle" message from the network.

and

Defines the idle signaling pattern to be sent to the PBX if the network trunk is out of service and signal sequence oos idle-only or signal sequence oos both is configured.

idle transmit

Defines the signaling pattern for identifying an "idle" message from the PBX.

oos receive

Defines the OOS signaling pattern to be sent to the PBX if the network trunk is out of service and signal sequence oos oos-only or signal sequence oos both is configured.

oos transmit

Defines the signaling pattern for identifying an OOS message from the PBX.

bit-pattern

The ABCD signaling bit pattern. Values are 0000 to 1111.


Defaults

idle receive

For near-end E&M—0000 (for T1) or 0001 (for E1)

For near-end FXO loop start—0101

For near-end FXO ground start—1111

For near-end FXS—0101

For near-end MELCAS—1101

idle transmit

For near-end E&M—0000

For near-end FXO—0101

For near-end FXS loop start—0101

For near-end FXS ground start—1111

For near-end MELCAS—1101

oos receive

For near-end E&M—1111

For near-end FXO loop start—1111

For near-end FXO ground start—0000

For near-end FXS loop start—1111

For near-end FXS ground start—0101

For near-end MELCAS—1111

oos transmit

No default signaling pattern is defined.


Command Modes

Voice-class configuration

Command History

Release
Modification

12.0(3)XG and 12.0(4)T

This command was introduced on the Cisco MC3810.

12.0(7)XK

Default signaling patterns were defined.


Usage Guidelines

This command defines the signaling patterns that are used to identify the idle and OOS states.

Before configuring the signaling pattern, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you will assign it to a dial peer.

To suppress voice packets whenever the transmit or receive trunk is in the idle state, use the idle receive and idle transmit commands in conjunction with the signal timing idle suppress-voice command.

To define the signaling bit patterns to be sent to the PBX when the trunk is out of service, use the idle receive and oos receive commands.

The oos receive pattern is the pattern sent to the PBX to indicate that the network trunk is out of service. The oos receive pattern is not used for pattern matching against the signaling packets received from the network.

To "busy out" a PBX if the network connection fails, set the oos receive pattern to match the seized state (busy); then set the signal timing oos timeout value. When the timeout expires and no signaling packets have been received, the router will send the idle receive and/or oos receive pattern to the PBX, depending on which pattern is specified by the signal sequence oos command.

Use the busy seized pattern only if the PBX does not have a pattern specifically intended to indicate an OOS state. If the PBX has a specific OOS pattern, use that pattern instead.

Examples

The following example restores default signaling bit patterns for the receive and transmit idle states:

router(config)# voice class permanent 10 
router(config-class)# signal keepalive 3 
router(config-class)# signal timing idle suppress-voice 
router(config-class)# no signal pattern idle receive  
router(config-class)# no signal pattern idle transmit  
router(config-class)# exit
router(config)# dial-peer voice 100 vofr
router(config-dial-peer)# voice-class permanent 10

The following example configures non-default signaling bit patterns for the receive and transmit idle states:

router(config)# voice class permanent 10 
router(config-class)# signal keepalive 3 
router(config-class)# signal timing idle suppress-voice 
router(config-class)# signal pattern idle receive 0101 
router(config-class)# signal pattern idle transmit 0101 
router(config-class)# exit
router(config)# dial-peer voice 100 vofr
router(config-dial-peer)# voice-class permanent 10

The following example restores default signaling bit patterns for the receive and transmit out-of-service states:

router(config)# voice class permanent 10 
router(config-class)# signal keepalive 3 
router(config-class)# signal timing idle suppress-voice 
router(config-class)# no signal pattern oos receive 
router(config-class)# no signal pattern oos transmit
router(config-class)# exit
router(config)# dial-peer voice 100 vofr
router(config-dial-peer)# voice-class permanent 10

The following example configures non-default signaling bit patterns for the receive and transmit out-of-service states:

router(config)# voice class permanent 10 
router(config-class)# signal keepalive 3 
router(config-class)# signal pattern oos receive 0001 
router(config-class)# signal pattern oos transmit 0001
router(config-class)# exit
router(config)# dial-peer voice 100 vofr
router(config-dial-peer)# voice-class permanent 10

Related Commands

Command
Description

dial-peer voice

Enters dial-peer configuration mode and specifies a dial-peer type.

signal keepalive

Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.

signal sequence oos

Specifies which signaling pattern is sent to the PBX when the far-end keepalive message is lost or AIS is received from the far end.

signal timing idle suppress-voice

Specifies the length of time before the router stops sending voice packets after a trunk goes into the idle state.

signal timing oos restart

Specifies that a permanent voice connection be torn down and restarted after the trunk has been OOS for a specified time.

signal timing oos slave-standby

Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time

signal timing oos suppress-all

Configures the router or concentrator to stop sending voice and signaling packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time.

signal timing oos suppress-voice

Configures the router or concentrator to stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time.

signal timing oos timeout

Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state.

signal timing idle suppress-voice

Specifies the length of time before voice traffic is stopped after a trunk goes into the idle state.

signal-type

Sets the signaling type to be used when connecting to a dial peer.

voice class permanent

Creates a voice class for a Cisco trunk or FRF.11 trunk.

voice-class permanent

Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.


signal sequence oos

To specify which signaling pattern is sent to the PBX when the far-end keepalive message is lost or AIS is received from the far end, use the signal sequence oos voice-class configuration command. Use the no form of this command to restore the default value.

signal sequence oos {no-action | idle-only | oos-only | both}
no signal sequence oos

Syntax Description

no-action

No signaling pattern is sent.

idle-only

Only the idle signaling pattern is sent.

oos-only

Only the out-of-service (OOS) signaling pattern is sent.

both

Both idle and OOS signaling patterns are sent. This is the default value.


Defaults

Both idle and OOS signal patterns are sent.

Command Modes

Voice-class configuration

Command History

Release
Modification

12.0(7)XK

This command was introduced on the Cisco MC3810.


Usage Guidelines

Before configuring the idle or OOS signal patterns to be sent, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you will assign it to a dial peer.

Use this command to specify which signaling pattern(s) to send. Use the signal pattern idle receive or the signal pattern oos receive command to define the bit patterns of the signaling patterns if other than the defaults.

Examples

The following example defines voice class 10, sets the signal sequence oos to send only the idle signal pattern to the PBX, and applies the voice class configuration to VoFR dial peer 100.

router(config)# voice class permanent 10
router(config-class)# signal keepalive 3 
router(config-class)# signal sequence oos idle-only 
router(config-class)# signal timing idle suppress-voice 5
router(config-class)# exit
router(config)# dial-peer voice 100 vofr
router(config-dial-peer)# voice-class permanent 10
router(config-dial-peer)# signal-type transparent

Related Commands

Command
Description

dial-peer voice

Enters dial-peer configuration mode and specifies a dial peer type.

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

Specifies the length of time before the router stops sending voice packets after a trunk goes into the idle state.

signal timing oos restart

Specifies that a permanent voice connection be torn down and restarted after the trunk has been OOS for a specified time.

signal timing oos slave-standby

Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time

signal timing oos suppress-all

Configures the router or concentrator to stop sending voice and signaling packets to the network if it detects a OOS signaling pattern from the PBX for a specified time.

signal timing oos suppress-voice

Configures the router or concentrator to stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time.

signal timing oos timeout

Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state.

signal-type

Sets the signaling type to be used when connecting to a dial peer.

voice class permanent

Creates a voice class for a Cisco trunk or FRF.11 trunk.

voice-class permanent

Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer.


signal timing idle suppress-voice

To specify the length of time before the router stops sending voice packets after a trunk goes into the idle state (no call in progress), use the signal timing idle suppress-voice voice-class configuration command. Use the no form of this command to restore the default value.

signal timing idle suppress-voice seconds
no signal timing idle suppress-voice

Syntax Description

seconds

Duration of the idle state in seconds before the transmission of voice packets is stopped. The range is 0 to 65535.


Defaults

The router or concentrator continues to send voice packets when the trunk is idle.

Command Modes

Voice-class configuration

Command History

Release
Modification

12.0(3)XG

This command was introduced on the Cisco MC3810.

12.0(7)XK

This command was modified to simplify the configuration process.


Usage Guidelines

Before configuring the signal timing idle suppress-voice timer, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. After you finish defining the voice class, you will assign it to a dial peer.

This command is used when the signal-type command is set to transparent in the dial peer for the Cisco trunk or FRF.11 trunk connection. When the router or concentrator stops sending voice packets after the specified time, signaling packets continue to be sent.

To detect an idle trunk state, the router or concentrator monitors both transmit and receive signaling for the idle transmit and idle receive signaling patterns. These can be configured by the signal pattern idle transmit or signal pattern idle receive command, or they can be the defaults. The default idle receive pattern is the idle pattern of the local voice port. The default idle transmit pattern is the idle pattern of the far-end voice port.

Examples

The following example defines voice class 10, sets the idle detection time to 5 seconds, configures the trunk to use the default transmit and receive idle signal patterns, and applies the voice class configuration to VoFR dial peer 100.

router(config)# voice class permanent 10
router(config-class)# signal keepalive 3 
router(config-class)# signal timing idle suppress-voice 5
router(config-class)# exit
router(config)# dial-peer voice 100 vofr
router(config-dial-peer)# voice-class permanent 10
router(config-dial-peer)# signal-type transparent

Related Commands

Command
Description

dial-peer voice

Enters dial-peer configuration mode and specifies a dial peer type.

signal keep