Voice over IP for the Cisco 3600 Series Commands

Table of Contents

Voice Over IP for the
Cisco 3600 Series Commands

Command Syntax Conventions
acc-qos
answer-address
codec
comfort-noise
connection
cptone
description
destination-pattern
dial-control-mib
dial-peer voice
dial-type
echo-cancel coverage
echo-cancel enable
expect-factor
fax-rate
icpif
impedance
input gain
ip precedence
ip udp checksum
music-threshold
non-linear
num-exp
operation
output attenuation
port
prefix
req-qos
ring frequency
ring number
session protocol
session target
show call active voice
show call history voice
show dial-peer voice
show dialplan incall number
show dialplan number
show num-exp
show voice port
shutdown (dial-peer configuration)
shutdown (voice-port configuration)
signal
snmp enable peer-trap poor-qov
snmp-server enable traps
snmp trap link-status
timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait
type
vad
voice-port

Voice Over IP for the
Cisco 3600 Series Commands

This chapter documents new commands. All other commands used with this feature are documented in the Cisco IOS Release 11.3 command reference documents.

The following new commands are used to configure and monitor Voice over IP:

  • acc-qos

  • answer-address

  • codec

  • comfort-noise

  • connection

  • cptone

  • description

  • destination-pattern

  • dial-control-mib

  • dial-peer voice

  • dial-type

  • echo-cancel coverage

  • echo-cancel enable

  • expect-factor

  • fax-rate

  • icpif

  • impedance

  • input gain

  • ip precedence

  • ip udp checksum

  • music-threshold

  • non-linear

  • num-exp

  • operation

  • output attenuation

  • port

  • prefix

  • req-qos

  • ring frequency

  • ring number

  • session-protocol

  • session-target

  • show call active voice

  • show call history voice

  • show dial-peer voice

  • show dialplan incall number

  • show dialplan number

  • show num-exp

  • show voice port

  • shutdown (dial-peer configuration)

  • shutdown (voice-port configuration)

  • signal

  • snmp enable peer-trap poor-qov

  • snmp-server enable traps

  • snmp trap link-status

  • timeouts initial

  • timeouts interdigit

  • timing clear-wait

  • timing delay-duration

  • timing delay-start

  • timing dial-pulse min-delay

  • timing digit

  • timing interdigit

  • timing pulse

  • timing pulse-interdigit

  • timing wink-duration

  • timing wink-wait

  • type

  • vad

  • voice-port

A subset of the commands listed are voice-port commands. Voice-port commands are supported on different voice signaling types, which vary depending on the platform. Table 4-1 lists the Cisco 3600 series voice port commands and the signaling types supported.


Table 4-1: Cisco 3600 Series Commands and Signaling Types Supported
Voice Port Command FXO FXS E&M

comfort-noise




connection




cptone

X
X
X

description

X
X
X

dial-type

X

X

echo-cancel coverage




echo-cancel enable




impedance

X
X
X

input gain

X
X
X

music-threshold




non-linear




operation



X

output attenuation

X
X
X

ring frequency


X

ring number

X


shutdown

X
X
X

signal

X
X
X

snmp trap link-status




timeouts initial




timeouts interdigit




timing




timing clear-wait



X

timing delay-duration



X

timing delay-start



X

timing dial-pulse min-delay



X

timing digit

X
X
X

timing inter-digit

X
X
X

timing pulse

X

X

timing pulse-inter-digit

X

X

timing wink-duration



X

timing wink-wait



X

type



X


Command Syntax Conventions

Table 4-2 describes the syntax used with the commands in this chapter.


Table 4-2: Command Syntax Guide

Convention Description

boldface font

Commands and keywords.

italic font

Command input that is supplied by you.

[     ]

Keywords or arguments that appear within square brackets are optional.

{ x | x | x }

A choice of keywords (represented by x) appears in braces separated by vertical bars. You must select one.

^ or Ctrl

Represent the key labeled Control. For example, when you read ^D or Ctrl-D, you should hold down the Control key while you press the D key.

screen font

Examples of information displayed on the screen.

boldface screen font

Examples of information that you must enter.

<     >

Nonprinting characters, such as passwords, appear in angled brackets.

[     ]

Default responses to system prompts appear in square brackets.

Note

Means reader take note. Notes contain helpful suggestions or references to additional information and material.

Caution

Means reader be careful. In this situation, you might do amoething that could result in equipemnt damage or loss of data.



Timesaver

Means the described action saves time. You can save time by performing the action described in the paragraph.



acc-qos

To generate an SNMP event if the quality of service for a dial peer drops below a specified level, use the acc-qos dial-peer configuration command. Use the no form of this command to use the default value for this feature.

acc-qos {best-effort | controlled-load | guaranteed-delay}
no acc-qos

Syntax Description

best-effort

Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation.

controlled-load

Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded.

guaranteed-delay

Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded.

Default

The default value is best-effort. Using the no form of this command is the same as the default.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the acc-qos dial-peer command to generate an SNMP event if the quality of service for specified dial peer drops below the specified level. When a dial peer is used, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.

To select the most appropriate value for this command, you need to be familiar with the amount of traffic this connection supports and what kind of impact you are willing to have on it. The Cisco IOS software generates a trap message when the bandwidth required to provide the selected quality of service is not available.

This command is only applicable to VoIP peers.

Example

The following example selects guaranteed-delay as the specified level below which an SNMP trap message will be generated:

dial-peer voice 10 voip
 acc-qos guaranteed-delay
Related Commands

You can use the master index or search online to find documentation of related commands.

req-qos

answer-address

To specify the full E.164 telephone number to be used to identify the dial peer of an incoming call, use the answer-address dial-peer configuration command. Use the no form of this command to disable this feature.

answer-address [+]string
no answer-address

Syntax Description

string

Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are:

· Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered.

· Plus sign (+), which is optionally used as the first digit to indicate an E.164 standard number.

· Comma (,), which inserts a pause between digits.

· Period (.), which matches any entered digit.

Default

The default value is enabled with a null string.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP network. Cisco IOS software identifies the dial peers of a call in one of two ways: either by identifying the interface through which the call is received or through the telephone number configured with the answer-address command. In the absence of a configured telephone number, the peer associated with the interface will be associated with the incoming call.

For calls coming in from a POTS interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer.

This command is applicable to both VoIP and POTS dial peers.


Note The Cisco IOS software does not check the validity of the E.164 telephone number; it will accept any series of digits as a valid number.
Example

The following example configures the E.164 telephone number, "555-9626" as the dial peer of an incoming call:

dial-peer voice 10 pots
 answer-address +5559626
Related Commands

You can use the master index or search online to find documentation of related commands.

destination-pattern
port
prefix

codec

To specify the voice coder rate of speech for a dial peer, use the codec dial-peer configuration command. Use the no form of this command to reset the default value for this command.

codec {g711alaw | g711ulaw | g729r8}
no codec

Syntax Description

g711alaw

G.711 A-Law 64,000 bits per second (bps).

g711ulaw

G.711 u-Law 64,000 bps.

g729r8

G.729 8000 bps.

Default

The default value for this command is g729r8.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the codec command to define a specific voice coder rate of speech for a dial peer.

For toll quality, use g711alaw or g711ulaw. These values provide high-quality voice transmission but use a significant amount of bandwidth. For almost toll quality (and a significant savings in bandwidth), use the g729r8 value.

If codec values for the VoIP peers of a connection do not match, the call will fail.

This command is only applicable to VoIP peers.

Example

The following example configures a voice coder rate that provides toll quality and uses a relatively high amount of bandwidth:

dial-peer voice 10 voip
 codec g711alaw

comfort-noise

To specify whether or not background noise should be generated, use the comfort-noise voice-port configuration command. Use the no form of this command to disable this feature.

comfort-noise
no comfort-noise


Syntax Description

This command has no arguments or keywords.

Default

The default value for this command is enabled.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the comfort-noise command to generate background noise to fill silent gaps during calls if VAD is activated. If comfort noise is not enabled, and VAD is enabled at the remote end of the connection, the user will hear dead silence when the remote party is not speaking.

The configuration of comfort noise only affects the silence generated at the local interface; it does not affect the use of VAD on either end of the connection, or the silence generated at the remote end of the connection.

Example

The following example enables background noise:

voice-port 1/0/0
 comfort-noise
Related Commands

You can use the master index or search online to find documentation of related commands.

vad

connection

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

connection plar string
no connection plar string

Syntax Description

plar

Specifies a private line auto ringdown (PLAR) connection. PLAR is handled by associating a peer directly with an interface; when an interface goes off-hook, the peer is used to set up the second call leg and conference them together without the caller having to dial any digits.

string

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

Default

The default value for this command is no connection.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the connection command to specify a connection mode for a specific interface. 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 calls coming in over this voice port. The destination dial peer is determined on the basis of this called number.

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.

Example

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

voice-port 1/0/0
 connection plar 5559262
Related Commands

You can use the master index or search online to find documentation of related commands.

session-protocol

cptone

To configure a voice call progress tone locale, use the cptone voice-port configuration command. Use the no form of this command to disable this feature.

cptone {australia | brazil | china | france | germany | japan | northamerica | unitedkingdom}
no cptone

Syntax Description

australia

Specifies an analog voice interface-related default tone, ring, and cadence setting for Australia.

brazil

Specifies an analog voice interface-related default tone, ring, and cadence setting for Brazil.

china

Specifies an analog voice interface-related default tone, ring, and cadence setting for China.

finland

Specifies an analog voice interface-related default tone, ring, and cadence setting for Finland.

france

Specifies an analog voice interface-related default tone, ring, and cadence setting for France.

germany

Specifies an analog voice interface-related default tone, ring, and cadence setting for Germany.

japan

Specifies an analog voice interface-related default tone, ring, and cadence setting for Japan.

northamerica

Specifies an analog voice interface-related default tone, ring, and cadence setting for North America.

unitedkingdom

Specifies an analog voice interface-related default tone, ring, and cadence setting for the United Kingdom.

Default

The default value for this command is northamerica.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the cptone command to specify a regional analog voice interface-related tone, ring, and cadence setting for a specified voice port. This command only affects the tones generated at the local interface. It does not affect any information passed to the remote end of a connection, or any tones generated at the remote end of a connection.

Example

The following example configures North America as the call progress tone locale:

voice-port 1/0/0
 cptone northamerica

description

To include a description of what this voice port is connected to, use the description voice-port configuration command. Use the no form of this command to disable this feature.

description string
no description


Syntax Description

string

Character string from 1 to 255 characters.

Default

The default for this command is enabled with a null string.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the description command to include descriptive text about this voice-port connection. This information is displayed when you enter a show command and does not affect the operation of the interface in any way.

Example

The following example identifies this voice port as being connected to the Purchasing department:

voice-port 1/0/0
 description purchasing_dept

destination-pattern

To specify either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer, use the destination-pattern dial-peer configuration command. Use the no form of this command to disable this feature.

destination-pattern [+]string
no destination-pattern

Syntax Description

string

Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are:

· Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered.

· Plus sign (+), which is optionally used as the first digit to indicate an E.164 standard number.

· Comma (,), which inserts a pause between digits.

· Period (.), which matches any entered digit.

Default

The default value for this command is enabled with a null string.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the destination-pattern command to define the E.164 telephone number for this dial peer. This pattern is used to match dialed digits to a dial peer. The dial peer is then used to complete the call.

This command is applicable to both VoIP and POTS dial peers.


Note The Cisco IOS software does not check the validity of the E.164 telephone number; it will accept any series of digits as a valid number.
Example

The following example configures the E.164 telephone number, "555-7922," for a dial peer:

dial-peer voice 10 pots
 destination-pattern +5557922
Related Commands

You can use the master index or search online to find documentation of related commands.

answer-address
prefix

dial-control-mib

To specify attributes for the call history table, use the dial-control-mib global configuration command.

dial-control-mib {max-size number | retain-timer number}

Syntax Description

max-size number

Specifies the maximum size of the call history table. Valid entries are from 0 to 500 table entries. A value of 0 will prevent any history from being retained.

retain-timer number

Specifies the length of time, in minutes, for entries in the call history table. Valid entries are from 0 to 2147483647 minutes. A value of 0 will prevent any history from being retained.

Default

The default call history table length is 50 table entries. The default retain timer is 15 minutes.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Example

The following example configures the call history table to hold 400 entries, with each entry remaining in the table for 10 minutes:

configure terminal
 dial-control-mib max-size 400
 dial-control-mib retain-timer 10

dial-peer voice

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

dial-peer voice number {voip | pots}

Syntax Description

number

Digit(s) defining a particular dial peer. Valid entries are from 1 to 2147483647.

voip

Indicates that this is a VoIP peer using voice encapsulation on the POTS network.

pots

Indicates that this is a POTS peer using Voice over IP encapsulation on the IP backbone.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

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.

Example

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

configure terminal
 dial-peer voice 10 pots
Related Commands

You can use the master index or search online to find documentation of related commands.

voice-port

dial-type

To specify the type of out-dialing for voice port interfaces, use the dial-type voice-port configuration command. Use the no form of this command to disable this feature.

dial-type {dtmf | pulse}
no dial-type

Syntax Description

dtmf

Specifies a touch-tone dialer.

pulse

Specifies a pulse dialer.

Default

The default value for this command is dtmf.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the dial-type command to specify an out-dialing type for an FXO or E&M voice port interface; this command is not applicable to FXS voice ports because they do not generate out-dialing. Voice ports can always detect dtmf and pulse signals. This command does not affect voice port dialing detection.

The dial-type command affects out-dialing as configured for the dial peer.

Example

The following example configures a voice port to support a touch-tone dialer:

voice-port 1/0/0
 dial-type dtmf

echo-cancel coverage

To adjust the size of the echo cancel, use the echo-cancel coverage voice-port configuration command. Use the no form of this command to reset this command to the default value.

echo-cancel coverage value
no echo-cancel coverage value

Syntax Description

value

Number of milliseconds the echo-canceller will cover on a given signal. Valid values are 16, 24, and 32.

Default

The default value for this command is 16.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the echo-cancel coverage command to adjust the coverage size of the echo canceller. This command enables cancellation of voice that is sent out the interface and received back on the same interface within the configured amount of time. If the local loop (the distance from the analog interface to the connected equipment producing the echo) is longer, the configured value of this command should be extended.

If you configure a longer value for this command, it will take the echo canceller longer to converge; in this case, the user might hear slight echo when the connection is initially set up. If the configured value for this command is too short, the user might hear some echo for the duration of the call because the echo canceller is not cancelling the longer delay echoes.

There is no echo or echo cancellation on the IP side of the connection.


Note This command is valid only if the echo cancel feature has been enabled. For more information, refer to the echo-cancel enable command.
Example

The following example adjusts the size of the echo canceller to 16 milliseconds:

voice-port 1/0/0
 echo-cancel enable
 echo-cancel coverage 16
Related Commands

You can use the master index or search online to find documentation of related commands.

echo-cancel enable

echo-cancel enable

To enable the echo cancel feature, use the echo-cancel enable voice-port configuration command. Use the no form of this command to disable this feature.

echo-cancel enable
no echo-cancel enable


Syntax Description

This command has no arguments or keywords.

Default

The default value for this command is enabled for all interface types.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The echo-cancel command enables cancellation of voice that is sent out the interface and is received back on the same interface. Disabling echo cancellation might cause the remote side of a connection to hear an echo. Because echo cancellation is an invasive process that can minimally degrade voice quality, this command should be disabled if it is not needed.

The echo-cancel command does not affect the echo heard by the user on the analog side of the connection.

There is no echo path for a 4-wire E&M interface. The echo canceller should be disabled for that interface type.


Note This command is valid only if the echo-cancel coverage command has been configured. For more information, refer to the echo-cancel coverage command.
Example

The following example enables the echo cancel feature for 16-millisecond echo coverage:

voice-port 1/0/0
 echo-cancel enable
 echo-cancel coverage 16
Related Commands

You can use the master index or search online to find documentation of related commands.

echo-cancel coverage
non-linear

expect-factor

To specify when the router will generate an alarm to the network manager, indicating that the expected quality of voice has dropped, use the expect-factor dial-peer configuration command. Use the no form of this command to reset the default value for this command.

expect-factor value
no expect-factor value

Syntax Description

value

Integers that represent the ITU specification for quality of voice as described in G.113. Valid entries are from 0 to 20, with 0 representing toll quality.

Default

The default value for this command is 10.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Voice over IP monitors the quality of voice received over the network. Use the expect-factor command to specify when the router will generate an SNMP trap to the network manager.

This command is only applicable to VoIP peers.

Example

The following example configures toll quality of voice when connecting to a dial peer:

dial-peer voice 10 voip
 expect-factor 0

fax-rate

To establish the rate at which a fax will be sent to the specified dial peer, use the fax-rate dial-peer configuration command. Use the no form of this command to reset the default value for this command.

fax-rate{2400 | 4800 | 7200 | 9600 | 14400 | disable | voice}
no fax-rate

Syntax Description

2400

Specifies a fax transmission speed of 2400 bits per second (bps).

4800

Specifies a fax transmission speed of 4800 bps.

7200

Specifies a fax transmission speed of 7200 bps.

9600

Specifies a fax transmission speed of 9600 bps.

14400

Specifies a fax transmission speed of 14,400 bps.

disable

Disables fax relay transmission capability.

voice

Specifies the highest possible transmission speed allowed by voice rate.

Default

The default value for this command is voice.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the fax-rate command to specify the fax transmission rate to the specified dial peer.

The values for this command apply only to the fax transmission speed and do not affect the quality of the fax itself. The higher values provide a faster transmission speed but monopolize a significantly larger portion of the available bandwidth. Slower transmission speeds use less bandwidth.

If the fax-rate command is set above the CODEC value in the same dial peer, the data sent over the network for fax transmission will be above the bandwidth reserved for RVSP. Because more network bandwidth will be monopolized by the fax transmission, we do not recommend setting the fax-rate value higher than the codec value. If the fax-rate value is set lower than the codec value, faxes will take longer to transmit but will use less bandwidth.

This command is only applicable to VoIP peers.

Example

The following example configures a facsimile rate of 9600 bps for faxes sent to a dial peer:

dial-peer voice 10 voip
 fax-rate 9600
Related Commands

You can use the master index or search online to find documentation of related commands.

codec

icpif

To specify the 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 0 to 55.

Default

The default value for this command is 30.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

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

impedance

To specify the terminating impedance of a voice port interface, use the impedance voice-port configuration command. Use the no form of this command to restore the default value.

impedance {600c | 600r | 900c | complex1 | complex2}
no impedance

Syntax Description

600c

Specifies 600 Ohms complex.

600r

Specifies 600 Ohms real.

900c

Specifies 900 Ohms complex.

complex1

Specifies Complex 1.

complex2

Specifies Complex 2.

Default

The default value for this command is 600.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the impedance command to specify the terminating impedance of an FXO voice-port interface. The impedance value selected needs to match the specifications from the specific telephony system to which it is connected. Different countries often have different standards for impedance. CO switches in the United States are predominantly 600r. PBXes in the United States are normally either 600r or 900c.

If the impedance is set incorrectly (if there is an impedance mismatch), there will be a significant amount of echo generated (which could be masked if the echo-cancel command has been enabled). In addition, gains might not work correctly if there is an impedance mismatch.

Configuring the impedance on a voice port will change the impedance on both voice ports of a VNM card. This voice port must be shut down and then opened for the new value to take effect.

This command is applicable to FXS, FXO, and E&M voice ports.

Example

The following example configures an FXO voice port for a terminating impedance of 600 Ohms:

impedance 600r

input gain

To configure a specific input gain value, use the input gain voice-port configuration command. Use the no form of this command to disable this feature.

input gain value
no input gain
value

Syntax Description

value

Specifies, in decibels, the amount of gain to be inserted at the receiver side of the interface. Acceptable value is any integer from -6 to 14.

Default

The default value for FXO, FXS, and E&M ports is 0.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

A system-wide loss plan must be implemented using both input gain and output attenuation commands. Other equipment (including PBXes) in the system must be taken into account when creating a loss plan. This default value for this command assumes that a standard transmission loss plan is in effect, meaning that normally, there must be -6 dB attenuation between phones. Connections are implemented to provide -6 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0.

Please note that you can't increase the gain of a signal going out into the PSTN, but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.

You can increase the gain of a signal coming in to the router. If the voice level is too low, you can increase the input gain.

Example

The following example configures a 3-decibel gain to be inserted at the receiver side of the interface:

input gain 3
Related Commands

You can use the master index or search online to find documentation of related commands.

output attenuation

ip precedence

To set IP precedence (priority) for packets sent by the dial peer, use the ip precedence dial-peer configuration command. Use the no form of this command to restore the default value for this command.

ip precedence number
no ip precedence

Syntax Description

number

Integer specifying the IP precedence value. Valid entries are 0 to 7. A value of 0 means that no precedence (priority) has been set.

Default

The default value for this command is 0.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the ip precedence command to configure the value set in the IP precedence field when voice data packets are sent over the IP network. This command should be used if the IP link utilization is high and the QoS for voice packets need to have a higher priority than other IP packets. The ip precedence command should also be used if RSVP is not enabled, and the user would like to give voice packets a higher priority over other IP data traffic.

This command is applicable to VoIP peers.

Example

The following example sets the IP precedence at 5:

dial-peer voice 10 voip
 ip precedence 5

ip udp checksum

To calculate the UDP checksum for voice packets transmitted by the dial peer, use the ip udp checksum dial-peer configuration command. Use the no form of this command to disable this feature.

ip udp checksum
no ip udp checksum


Syntax Description

This command has no arguments or keywords.

Default

The default value for this command is disabled.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the ip udp checksum command to enable UDP checksum calculation for each of the outbound voice packets. This command is disabled by default to speed up the transmission of the voice packets. If you suspect that the connection has a high error rate, you should enable ip udp checksum to prevent bad voice packets forwarded to the DSP.

This command is applicable to VoIP peers.

Example

The following example calculates the UDP checksum for voice packets transmitted by this dial peer:

dial-peer voice 10 voip
 ip udp checksum

music-threshold

To specify the threshold for on-hold music for a specified voice port, use the music-threshold voice-port configuration command. Use the no form of this command to disable this feature.

music-threshold number
no music-threshold number

Syntax Description

number

Specifies the on-hold music threshold in decibels. Valid entries are any integer from -70 to -30.

Default

The default value for this command is -38.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the music-threshold command to specify the decibel level of music played when calls are put on hold. This command tells the firmware to pass steady data above the specified level. It only affects the operation of VAD when receiving voice.

If the value for this command is set too high, VAD will interpret music-on-hold as silence, and the remote end will not hear the music. If the value for this command is set too low, VAD will compress and pass silence when the background is noisy, creating unnecessary voice traffic.

Example

The following sets the decibel threshold to -35 for the music played when calls are put on hold:

voice port 1/0/0
 music-threshold -35

non-linear

To enable non-linear processing in the echo canceller, use the non-linear voice-port configuration command. Use the no form of this command to disable this feature.

non-linear
no non-linear


Syntax Description

This command has no arguments or keywords.

Default

The default for this command is enabled for all voice-port types.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

This command is associated with the echo canceller operation. The echo-cancel enable command must be enabled for the non-linear command to take effect. Use the non-linear command to shut off any signal if no near-end speech is detected.

Enabling the non-linear command normally improves performance, although some users might perceive truncation of consonants at the end of sentences when this command is enabled.

This feature is also generally known as residual echo suppression.

Example

The following example enables non-linear call processing:

voice-port 1/0/0
 non-linear
Related Commands

You can use the master index or search online to find documentation of related commands.

echo-cancel enable

num-exp

To define how to expand an extension number into a particular destination pattern, use the num-exp global configuration command.

num-exp extension-number expanded-number

Syntax Description

extension-number

Digit(s) defining an extension number for a particular dial peer.

expanded-number

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

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the num-exp global configuration command to define how to expand a particular set of numbers (for example, an extension number) into a particular 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 7-digit numbers to numbers containing less than 7 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—meaning that if you want to replace 4 numbers in an extension with wildcards, type in 4 periods.

Examples

The following example expands the extension number 55541 to be expanded to 14085555541:

num-exp 65541 14085555541
 

The following example shows how to expand all 5-digit extensions beginning with 5 to append the following numbers at the beginning of the extension number 1408555:

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

operation

To select a specific cabling scheme for E&M ports, use the operation voice-port configuration command. Use the no form of this command as an alternative method of configuring 2-wire operation.

operation {2-wire | 4-wire}
no operation {2-wire | 4-wire}

Syntax Description

2-wire

Specifies a 2-wire E&M cabling scheme.

4-wire

Specifies a 4-wire E&M cabling scheme.

Default

2-wire operation

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The operation command will only affect voice traffic. Signaling is independent of 2-wire versus 4-wire settings. If the wrong cable scheme is specified, the user might get voice traffic in only one direction.

Configuring the operation command on a voice port changes the operation of both voice ports on a VPM card. The voice port must be shut down and then opened again for the new value to take effect.

This command is not applicable to FXS or FXO interfaces because those are, by definition, two-wire interfaces.

Example

The following example specifies that an E&M port uses a 4-wire cabling scheme:

voice-port 1/0/0
 operation 4-wire

output attenuation

To configure a specific output attenuation value, use the output attenuation voice-port configuration command. Use the no form of this command to disable this feature.

output attenuation value
no output attenuation


Syntax Description

value

Specifies, in decibels, the amount of attenuation at the transmit side of the interface. Acceptable value is any integer from 0 to 14. The default value for FXO, FXS, and E&M ports is 0.

Default

The default value for FXO, FXS, and E&M ports is 0.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

A system-wide loss plan must be implemented using both input gain and output attenuation commands. Other equipment (including PBXes) in the system must be taken into account when creating a loss plan. This default value for this command assumes that a standard transmission loss plan is in effect, meaning that normally, there must be -6 dB attenuation between phones. Connections are implemented to provide -6 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0.

Please note that you can't increase the gain of a signal going out into the PSTN, but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.

Example

The following example configures a 3-decibel gain to be inserted at the transmit side of the interface:

voice-port 1/0/0
 output attenuation 3
Related Commands

You can use the master index or search online to find documentation of related commands.

input gain

port

To associate a dial peer with a specific voice-port, use the port dial-peer configuration command. Use the no form of this command to cancel this association.

port slot-number/subunit-number/port
no port

Syntax Description

slot-number/

Specifies the slot number in the Cisco router where the voice interface card is installed. Valid entries are from 0 to 3, depending on the slot where it has been installed.

subunit-number/

Specifies the subunit on the voice interface card where the voice port is located. Valid entries are 0 or 1.

port

Specifies the voice port. Valid entries are 0 or 1.

Default

No port is configured.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the port configuration command to associate the designated voice port with the selected dial peer.

This command is used for calls incoming from a telephony interface to select an incoming dial peer and for calls coming from the VoIP network to match a port with the selected outgoing dial peer.

This command is applicable only to POTS peers.

Example

The following example associates a dial peer with voice port 1, which is located on subunit 0, and accessed through port 0:

dial-peer voice 10 pots
 port 1/0/0

prefix

To specify the prefix of the dialed digits for this dial peer, use the prefix dial-peer configuration command. Use the no form of this command to disable this feature.

prefix string
no prefix

Syntax Description

string

Integers representing the prefix of the telephone number associated with the specified dial peer. Valid numbers are 0 through 9, and a comma (,). Use a comma to include a pause in the prefix.

Default

The default for this command is a null string.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the prefix command to specify a prefix for a specific dial peer. When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer.

If you want to configure different prefixes for dialed numbers on the same interface, you need to configure different dial peers.

This command is applicable only to POTS peers.

Example

The following example specifies a prefix of "9" and then a pause:

dial-peer voice 10 pots
 prefix 9,
Related Commands

You can use the master index or search online to find documentation of related commands.

answer-address
destination-pattern

req-qos

To specify the desired quality of service to be used in reaching a specified dial peer, use the req-qos dial-peer configuration command. Use the no form of this command to restore the default value for this command.

req-qos {best-effort | controlled-load | guaranteed-delay}
no req-qos

Syntax Description

best-effort

Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation.

controlled-load

Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded.

guaranteed-delay

Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded.

Default

The default value for this command is best-effort. The no form of this command restores the default value.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you enter this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.

This command is applicable only to VoIP peers.

Example

The following example configures guaranteed-delay as the desired (requested) quality of service to a dial peer:

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

You can use the master index or search online to find documentation of related commands.

acc-qos

ring frequency

To specify the ring frequency for a specified FXS voice port, use the ring frequency voice-port configuration command. Use the no form of this command to reset the default value for this command.

ring frequency number
no ring frequency

Syntax Description

number

Specifies the ring frequency (Hertz) used in the FXS interface. Valid entries are 25 and 50.

Default

The default value for this command is 25 Hertz.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the ring frequency command to select a specific ring frequency for an FXS voice port. Use the no form of this command to reset the default value for this command, which is 25 Hertz. The ring frequency you select must match the connected equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring frequency is usually country-dependent and you should take into account the appropriate ring frequency for your area before configuring this command.

This command does not affect ringback, which is the ringing a user hears when placing a remote call.

Example

The following example configures the ring frequency for 50 Hertz:

voice-port 1/0/0
 ring frequency 50
Related Commands

You can use the master index or search online to find documentation of related commands.

ring number

ring number

To specify the number of rings for a specified FXO voice port, use the ring number voice-port configuration command. Use the no form of this command to reset the default value for this command.

ring number number
no ring number number

Syntax Description

number

Specifies the number of rings detected before answering the call. Valid entries are numbers from 1 to 10.

Default

The default value is 1 ring.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the ring number command to set the maximum number of rings to be detected before answering a call over an FXO voice port. Use the no form of this command to reset the default value, which is 1 ring.

Normally, this command should be set to the default so that incoming calls are answered quickly. If you have other equipment available on the line to answer incoming calls, you might want to set the value higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if the equipment online did not answer the incoming call in the configured number of rings.

This command is not applicable to FXS or E&M interfaces because they do not receive ringing to receive a call.

Example

The following example sets 5 rings as the maximum number of rings to be detected before closing a connection over this voice port:

voice port 1/0/0
 ring number 5
Related Commands

You can use the master index or search online to find documentation of related commands.

ring frequency

session protocol

To establish a session protocol for calls between the local and remote routers via the packet network, use the session protocol dial-peer configuration command. Use the no form of this command to reset the default value for this command.

session protocol cisco
no session protocol

Syntax Description

cisco

Specifies Cisco Session Protocol.

Default

The default value for this command is cisco.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

For this release, Cisco Session Protocol (cisco) is the only applicable session protocol. This command is applicable only to VoIP peers.

Example

The following example selects Cisco Session Protocol as the session protocol:

dial-peer voice 10 voip
 session protocol cisco
Related Commands

You can use the master index or search online to find documentation of related commands.

session target

session target

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

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

Syntax Description

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 1 of the following 3 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.

· $u$.—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.

Default

The default for this command is enabled with no IP address or domain name defined.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

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.

Example

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 1310555. 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 1310555....
 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 13105551111. 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 13105551111
 session target dns:$d$.cisco.com
Related Commands

You can use the master index or search online to find documentation of related commands.

destination-pattern
session protocol

show call active voice

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

show call active voice

Syntax Description

This command contains no arguments or keywords.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the show call active voice privileged EXEC command to display the contents of the active call table, which shows all of the calls currently connected through the router.

For each call, there are 2 call legs, usually a POTS call leg and a VoIP call leg. A call leg is a discrete segment of a call connection that lies between 2 points in the connection. Each dial peer creates a call leg, as shown in Figure 4-1.


Figure 4-1: Call Legs Example


These 2 call legs are associated by the connection ID. The connection ID is global across the voice network, so that you can associate 2 call legs on one router with 2 call legs on another router, thereby providing an end-to-end view of a call.

Sample Display

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

sloth# 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=

Table 4-3 provides an alphabetical listing of the fields in this output and a description of each field.


Table 4-3: Show Call Active Voice Field Descriptions
Field Description

ACOM Level

Current ACOM level for the call. This value is 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.

GapFillWith Silence

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.

GapFillWith Redundancy

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

You can use the master index or search online to find documentation of related commands.

show call history voice
show dial-peer voice
show num-exp
show voice port

show call history voice

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

show call history voice last number

Syntax Description

last number

Displays the last calls connected, where the number of calls displayed is defined by the argument number. A valid entry for the argument number is any number from 1 to 2147483647.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all calls connected through this router in descending time order since Voice over IP was enabled. 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.

Sample Display

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

sloth# 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=

Table 4-4 provides an alphabetical listing of the fields in this output and a description of each field.


Table 4-4: 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.

Session Protocol

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

Session Target

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

You can use the master index or search online to find documentation of related commands.

show call active voice
show dial-peer voice
show num-exp
show voice port

show dial-peer voice

To display configuration information for dial peers, use the show dial-peer voice privileged EXEC command.

show dial-peer voice [number]

Syntax Description

number

Displays configuration for the dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the show dial-peer voice privileged EXEC command to display the configuration for all VoIP and POTS dial peers configured for the router. To show configuration information for only one specific dial peer, use the argument number to identify the dial peer.

Sample Display

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

sloth# show dial-peer voice 1

VoiceEncapPeer1
        tag = 1, dest-pat = \Q+14085551000',
        answer-address = \Q',
        group = 0, Admin state is up, Operation state is down
        Permission is Both,
        type = pots, prefix = \Q',
        session-target = \Q', voice-port =
        Connect Time = 0, Charged Units = 0
        Successful Calls = 0, Failed Calls = 0
        Accepted Calls = 0, Refused Calls = 0
        Last Disconnect Cause is ""
        Last Disconnect Text is ""
        Last Setup Time = 0 
 

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

sloth# show dial-peer voice 10

VoiceOverIpPeer10
        tag = 10, dest-pat = \Q',
        incall-number = \Q+14087',
        group = 0, Admin state is up, Operation state is down
        Permission is Answer, 
        type = voip, session-target = \Q',
        sess-proto = cisco, req-qos = bestEffort, 
        acc-qos = bestEffort, 
        fax-rate = voice, codec = g729r8,
        Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled, 
        Connect Time = 0, Charged Units = 0
        Successful Calls = 0, Failed Calls = 0
        Accepted Calls = 0, Refused Calls = 0
        Last Disconnect Cause is ""
        Last Disconnect Text is ""
        Last Setup Time = 0
 

Table 4-5 explains the fields contained in both of these examples.


Table 4-5: Show Dial-Peer Voice Field Descriptions
Field Description

AcceptedCalls

Number of calls from this peer accepted since system startup.

acc-qos

Lowest acceptable quality of service configured for calls for this peer.

Admin state

Administrative state of this peer.

Charged Units

Total number of charging units applying to this peer since system startup.The unit of measure is in hundredths of seconds.

codec

Default voice coder rate of speech for this peer.

Connect Time

Accumulated connect time to the peer since system startup for both incoming and outgoing calls. The unit of value is in hundredths of seconds.

dest-pat

Destination pattern (telephone number) for this peer.

Expect factor

User-requested Expectation Factor of voice quality for calls via this peer.

fax-rate

Fax transmission rate configured for this peer.

Failed Calls

Number of failed call attempts to this peer since system startup.

group

Group number associated with this peer.

ICPIF

Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer.

incall-number

Full E.164 telephone number to be used to identify the dial peer.

Last Disconnect Cause

Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface.

Last Disconnect Text

ASCII text describing the reason for the last call termination.

Last Setup Time

Value of the System Up Time when the last call to this peer was started.

Operation state

Operational state of this peer.

Permission

Configured permission level for this peer.

Poor QOV Trap

Whether Poor Quality of Voice trap messages have been enabled or disabled.

Refused Calls

Number of calls from this peer refused since system startup.

req-qos

Configured requested quality of service for calls for this dial peer.

session-target

Session target of this peer.

sess-proto

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

Successful Calls

Number of completed calls to this peer.

tag

Unique dial-peer ID number.

VAD

Whether or not voice activation detection (VAD) is enabled for this dial peer.



Related Commands

You can use the master index or search online to find documentation of related commands.

show call active voice
show call-history voice
show num-exp
show voice port

show dialplan incall number

To pair different voice ports and telephone numbers together for troubleshooting, use the show dialplan incall number privileged EXEC command.

show dialplan incall slot-number/subunit-number/port number dial string

Syntax Description

slot-number/

Specifies the slot number in the Cisco router where the voice network module is installed. Valid entries are from 0 to 3, depending on the voice interface card you have installed.

subunit-number/

Specifies the subunit on the voice network module where the voice port is located. Valid entries are 0 or 1.

port

Specifies the voice port. Valid entries are 0 or 1.

dial string

Specifies a particular destination pattern (telephone number).

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Occasionally, an incoming call cannot be matched to a dial peer in the dial-peer database. One reason this might occur is that the specified destination cannot be reached via the voice interface through which the incoming call came. Use the show dialplan incall number command as a troubleshooting method to resolve the call destination by pairing voice ports and telephone numbers together until there is a match.

Example

The following example tests whether the telephone extension 57681 can be reached through voice port 1/0/1:

show dialplan incall 1/0/1 number 57681
Related Commands

You can use the master index or search online to find documentation of related commands.

show dialplan number

show dialplan number

To show which dial peer is reached when a particular telephone number is dialed, use the show dial plan number privileged EXEC command.

show dial plan number dial string

Syntax Description

dial string

Specifies a particular destination pattern (telephone number).

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Example

The following example displays the dial peer associated with the destination pattern of 54567:

show dialplan number 54567
Related Commands

You can use the master index or search online to find documentation of related commands.

show dialplan incall number

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

Displays number expansion for the specified dialed number.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

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.

Sample Display

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

sloth# show num-exp

Dest Digit Pattern = '0...'     Translation = '+14085550...'
Dest Digit Pattern = '1...'     Translation = '+14085551...'
Dest Digit Pattern = '3..'      Translation = '+140855503..'
Dest Digit Pattern = '4..'      Translation = '+140855504..'
Dest Digit Pattern = '5..'      Translation = '+140855505..'
Dest Digit Pattern = '6....'    Translation = '+1408555....'
Dest Digit Pattern = '7....'    Translation = '+1408555....'
Dest Digit Pattern = '8...'     Translation = '+14085558...'
 

Table 4-6 explains the fields in the sample output.


Table 4-6: Show Dial-Peer 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

You can use the master index or search online to find documentation of related commands.

show call active voice
show call history voice
show dial-peer voice
show voice port

show voice port

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

show voice port slot-number/subunit-number/port

Syntax Description

slot-number/

Specifies the slot number in the Cisco router where the voice interface card is installed. Valid entries are from 0 to 3, depending on the slot where it has been installed.

subunit-number/

Specifies the subunit on the voice interface card where the voice port is located. Valid entries are 0 or 1.

port

Specifies the voice port. Valid entries are 0 or 1.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

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

Sample Display

The following is sample output from the show voice port command for an E&M voice port:

sloth# 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 voice port:

sloth# 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 
 
 

Table 4-7 explains the fields in the sample output.


Table 4-7: Show Voice Port Field Descriptions
Field Description

Administrative State

Administrative state of the voice port.

Alias

User-supplied alias for this voice port.

Clear Wait Duration Timing

Time of inactive seizure signal to declare call cleared.

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.

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.

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.

Number of signaling protocol errors

Number of signaling protocol errors.

Non-Linear Processing

Whether or not Non-Linear Processing is enabled for this port.

Operations State

Operation state of the port.

Operation Type

Operation of the E&M signal: 2-wire or 4-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).

Regional Tone

Configured regional tone for this interface.

Ring Active Status

Ring active indication.

Ring Frequency

Configured ring frequency for this interface.

Ring Ground Status

Ring ground indication.

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,

Wink Duration Timing

Maximum wink duration for wink start signaling.

Wink Wait Duration Timing

Maximum wink wait duration for wink start signaling.



Related Commands

You can use the master index or search online to find documentation of related commands.

show call active voice
show call history voice
show dial-peer voice
show num-exp

shutdown (dial-peer configuration)

To change the administrative state of the selected dial peer from up to down, use the shutdown dial-peer configuration command. Use the no form of this command to change the administrative state of this dial peer from down to up.

shutdown
no shutdown


Syntax Description

This command has no arguments or keywords.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

When a dial peer is shut down, you cannot initiate calls to that peer. This command is applicable to both VoIP and POTS peers.

Example

The following example changes the administrative state of voice telephony dial peer 10 to down:

configure terminal
 dial-peer voice 10 pots
 shutdown
 

shutdown (voice-port configuration)

To take the voice ports for a specific voice interface card offline, use the shutdown voice-port configuration command. Use the no form of this command to put the ports back in service.

shutdown
no shutdown


Syntax Description

This command has no arguments or keywords.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

When you enter the shutdown command, all ports on the voice interface card are disabled. When you enter the no shutdown command, all ports on the voice interface card are enabled. A telephone connected to an interface will hear dead silence when a port is shut down.

Example

The following example takes voice port 1/1/0 offline:

configure terminal 
 voice port 1/1/0
 shutdown

Note The preceding configuration example will shut down both voice ports 1/1/0 and 1/1/1.

signal

To specify the type of signaling for a voice port, use the signal voice-port configuration command. Use the no form of this command to restore the default value for this command.

signal {loop-start | ground-start | wink-start | immediate | delay-dial}
no signal

Syntax Description

loop-start

Specifies Loop Start signaling. Used for FXO and FXS interfaces. With Loop Start signaling only one side of a connection can hang up. This is the default setting for FXO and FXS voice ports.

ground-start

Specifies Ground Start signaling. Used for FXO and FXS interfaces. Ground Start allows both sides of a connection to place a call and to hang up.

wink-start

Indicates that the calling side seizes the line by going off-hook on its E lead then waits for a short off-hook "wink" indication on its M lead from the called side before sending address information as DTMF digits. Used for E&M tie trunk interfaces. This is the default setting for E&M voice ports.

immediate

Indicates that the calling side seizes the line by going off-hook on its E lead and sends address information as DTMF digits. Used for E&M tie trunk interfaces.

delay-dial

Indicates that the calling side seizes the line by going off-hook on its E lead. After a timing interval, the calling side looks at the supervision from the called side. If the supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise, the calling side waits until the called side goes on-hook and then starts sending address information. Used for E&M tie trunk interfaces.

Default

The default value is loop-start for FXO and FXS interfaces. The default value is wink-start for E&M interfaces.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Configuring the signal command for an FXS or FXO voice port will change the signal value for both voice ports on a VPM card.


Note If you change the signal type for an FXO voice port, you need to move the appropriate jumper in the voice interface card of the voice network module. For more information about the physical characteristics of the voice network module, refer to the Voice Network Module and Voice Interface Card Configuration Note that came with your voice network module

Configuring this command for an E&M voice port will change only the signal value for the selected voice port. In either case, the voice port must be shut down and then activated before the configured values will take effect.

Some PBXes will miss initial digits if the E&M voice port is configured for Immediate signaling. If this occurs, use Delay-Dial signaling instead. Some non-Cisco devices have a limited number of DTMF receivers. This type of equipment must delay the calling side until a DTMF receiver is available.

Example

The following example configures Ground Start signaling, which means that both sides of a connection can place a call and hang up, as the signaling type for a voice port:

configure terminal
 voice-port 1/1/1
 signal ground-start

snmp enable peer-trap poor-qov

To generate poor quality of voice notification for applicable calls associated with VoIP dial peers, use the snmp enable peer-trap poor-qov dial-peer configuration command. Use the no form of this command to disable this feature.

snmp enable peer-trap poor-qov
no snmp enable peer-trap poor-qov


Syntax Description

This command has no arguments or keywords.

Default

Disabled

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the snmp enable peer-trap poor qov command to generate poor quality of voice notifications for applicable calls associated with this dial peer. If you have an SNMP manager that will use SNMP messages when voice quality drops, you might want to enable this command. Otherwise, you should disable this command to reduce unnecessary network traffic.

This command is applicable only to VoIP peers.

Example

The following example enables poor quality of voice notifications for calls associated with VoIP dial peer 10:

dial-peer voice 10 voip
 snmp enable peer-trap poor-qov
Related Commands

You can use the master index or search online to find documentation of related commands.

snmp-server enable trap voice poor-qov
snmp trap link-status

snmp-server enable traps

To enable the router to send SNMP traps, use the snmp-server enable traps global configuration command. Use the no form of this command to disable SNMP traps.

snmp-server enable traps [trap-type] [trap-option]
no snmp-server enable traps [trap-type] [trap-option]

Syntax Description

trap-type

(Optional) Type of trap to enable. If no type is specified, all traps are sent (including the envmon and repeater traps). The trap type can be one of the following keywords:

· bgp—Sends Border Gateway Protocol (BGP) state change traps.

· config—Sends configuration traps.

· entity—Sends Entity MIB modification traps.

· envmon—Sends Cisco enterprise-specific environmental monitor traps when an environmental threshold is exceeded. When the envmon keyword is used, you can specify a trap-option value.

· frame-relay—Sends Frame Relay traps.

· isdn—Sends Integrated Services Digital Network (ISDN) traps. When the isdn keyword is used on Cisco 1600 series routers, you can specify a trap-option value.

· repeater—Sends Ethernet hub repeater traps. When the repeater keyword is selected, you can specify a trap-option value.

· rtr—Sends response time reporter (RTR) traps.

· snmp—Sends Simple Network Management Protocol (SNMP) traps. When the snmp keyword is used, you can specify a trap-option value.

· syslog—Sends error message traps (Cisco Syslog MIB). Specify the level of messages to be sent with the logging history level command.

· voice—Sends SNMP poor quality of voice traps, when used with the qov trap-option.

trap-option

(Optional) When the envmon keyword is used, you can enable a specific environmental trap type, or accept all trap types from the environmental monitor system. If no option is specified, all environmental types are enabled. The option can be one or more of the following keywords: voltage, shutdown, supply, fan, and temperature.

When the isdn keyword is used on Cisco 1600 series routers, you can specify the call-information keyword to enable an SNMP ISDN call information trap for the ISDN MIB subsystem, or you can specify the isdnu-interface keyword to enable an SNMP ISDN U interface trap for the ISDN U interface MIB subsystem.

When the repeater keyword is used, you can specify the repeater option. If no option is specified, all repeater types are enabled. The option can be one or more of the following keywords:

· health—Enables IETF Repeater Hub MIB (RFC 1516) health trap.

· resetEnables IETF Repeater Hub MIB (RFC 1516) reset trap.

When the snmp keyword is used, you can specify the authentication option to enable SNMP Authentication Failure traps. (The snmp-server enable traps snmp authentication command replaces the snmp-server trap-authentication command.) If no option is specified, all SNMP traps are enabled.

When the voice keyword is used, you can enable SNMP poor quality of voice traps by using the qov option.

Defaults

This command is disabled by default. No traps are enabled.

Some trap types cannot be controlled with this command. These traps are either always enabled or enabled by some other means. For example, the linkUpDown messages are disabled by the no snmp trap link-status command.

If you enter this command with no keywords, the default is to enable all trap types.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.1.

This command is useful for disabling traps that are generating a large amount of uninteresting or useless noise.

If you do not enter an snmp-server enable traps command, no traps controlled by this command are sent. In order to configure the router to send these SNMP traps, you must enter at least one snmp-server enable traps command. If you enter the command with no keywords, all trap types are enabled. If you enter the command with a keyword, only the trap type related to that keyword is enabled. In order to enable multiple types of traps, you must enter a separate snmp-server enable traps command for each trap type and option.

The snmp-server enable traps command is used in conjunction with the snmp-server host command. Use the snmp-server host command to specify which host or hosts receive SNMP traps. In order to send traps, you must configure at least one snmp-server host command.

For a host to receive a trap controlled by this command, both the snmp-server enable traps command and the snmp-server host command for that host must be enabled. If the trap type is not controlled by this command, just the appropriate snmp-server host command must be enabled.

The trap types used in this command all have an associated MIB object that allows them to be globally enabled or disabled. Not all of the trap types available in the snmp-server host command have notificationEnable MIB objects, so some of these cannot be controlled using the snmp-server enable traps command.

Examples

The following example enables the router to send SNMP poor quality of voice traps:

configure terminal 
 snmp-server enable trap voice poor-qov

The following example enables the router to send all traps to the host myhost.cisco.com using the community string public:

snmp-server enable traps
snmp-server host myhost.cisco.com public
 

The following example enables the router to send Frame Relay and environmental monitor traps to the host myhost.cisco.com using the community string public:

snmp-server enable traps frame-relay
snmp-server enable traps envmon temperature
snmp-server host myhost.cisco.com public
 

The following example will not send traps to any host. The BGP traps are enabled for all hosts, but the only traps enabled to be sent to a host are ISDN traps.

snmp-server enable traps bgp
snmp-server host bob public isdn
Related Commands

You can use the master indexes or search online to find documentation of related commands.

snmp enable peer-trap peer-qov
snmp-server host
snmp-server trap-source
snmp trap illegal-address
snmp trap link-status

snmp trap link-status

To enable Simple Network Management Protocol (SNMP) trap messages to be generated when this voice port is brought up or down, use the snmp trap link-status voice-port configuration command. Use the no form of this command to disable this feature.

snmp trap link-status
no snmp trap link-status


Syntax Description

This command contains no arguments or keywords.

Default

The default for this command is enabled.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the snmp trap link-status command to enable SNMP trap messages (linkup and linkdown) to be generated whenever this voice port is brought online or offline.

If you are managing the equipment with an SNMP manager, this command should be enabled. Enabling link-status messages will allow the SNMP manager to learn of a status change without polling the equipment. If you are not using an SNMP manager, this command should be disabled to avoid unnecessary network traffic.

Example

The following example enables SNMP trap messages for voice-port 2/1/0:

voice-port 2/1/0
 snmp trap link-stat
Related Commands

You can use the master index or search online to find documentation of related commands.

smnp enable peer-trap poor-qov
snmp-server enable trap poor-qov

timeouts initial

To configure the initial digit timeout value for a specified voice port, use the timeouts initial voice-port configuration command. Use the no form of this command to restore the default value for this command.

timeouts initial seconds
no timeouts initial seconds

Syntax Description

initial seconds

Specifies the initial timeout duration in seconds. Valid entries are any integer from 0 to 120.

Default

The default value is 10 seconds.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the timeouts initial command to specify the number of seconds the system will wait for the caller to input the first digit of the dialed digits. The timeouts initial timer is activated when the call is accepted and is deactivated when the caller inputs the first digit. If the configured timeout value is exceeded, the caller is notified through the appropriate tone and the call is terminated.

To disable the timeouts initial timer, set the seconds value to 0.

Example

The following example sets the initial digit timeout value to 15 seconds:

voice-port 1/0/0
 timeouts initial 15
Related Commands

You can use the master index or search online to find documentation of related commands.

timeouts interdigit
timing

timeouts interdigit

To configure the interdigit timeout value for a specified voice port, use the timeouts interdigit voice-port configuration command. Use the no form of this command to restore the default value for this command.

timeouts interdigit seconds
no timeouts interdigit seconds

Syntax Description

seconds

Specifies the interdigit timeout duration in seconds. Valid entries are any integer from 0 to 120.

Default

The default value is 10 seconds.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the timeouts interdigit command to specify the number of seconds the system will wait (after the caller has input the initial digit) for the caller to input a subsequent digit of the dialed digits. The timeouts interdigit timer is activated when the caller inputs a digit and restarted each time the caller inputs another digit until the destination address is identified. If the configured timeout value is exceeded before the destination address is identified, the caller is notified through the appropriate tone and the call is terminated.

To disable the timeouts interdigit timer, set the seconds value to 0.

Example

The following example sets the interdigit timeout value for 15 seconds:

voice-port 1/0/0
 timeouts interdigit 15
Related Commands

You can use the master index or search online to find documentation of related commands.

timeouts initial
timing

timing clear-wait

To indicate the minimum amount of time between the inactive seizure signal and the call being cleared for a specified voice port, use the timing clear-wait voice-port configuration command. Use the no form of this command to reset the default value.

timing clear-wait milliseconds
no timing clear-wait milliseconds

Syntax Description

milliseconds

Minimum amount of time, in milliseconds, between the inactive seizure signal and the call being cleared. Valid entries on the Cisco 3600 series are numbers from 200 to 2000. Supported on E&M ports only.

Default

400 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Examples

The following example configures the clear-wait duration on a Cisco 3600 series voice port to 300 milliseconds:

voice-port 1/0/0
 timing clear-wait 300
 
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing delay-duration

To specify the delay signal duration for a specified voice port, use the timing delay-duration voice-port configuration command. Use the no form of this command to reset the default value.

timing delay-duration milliseconds
no timing delay-duration milliseconds

Syntax Description

milliseconds

Delay signal duration for delay dial signaling, in milliseconds. Valid entries are numbers from 100 to 5000. Supported on E&M ports only.

Default

2000 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call direction for the timing delay-duration command is out.

Examples

The following example configures the delay signal duration on a Cisco 3600 series voice port to 3000 milliseconds:

voice-port 1/0/0
 timing delay-duration 3000
 
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing delay-start

To specify the minimum delay time from outgoing seizure to out-dial address for a specified voice port, use the timing delay-start voice-port configuration command. Use the no form of this command to reset the default value.

timing delay-start milliseconds
no timing delay-start milliseconds

Syntax Description

milliseconds

Minimum delay time, in milliseconds, from outgoing seizure to out-dial address. Valid entries are numbers from 20 to 2000. Supported on E&M ports only.

Default

300 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call direction for the timing delay-start command is out.

Examples

The following example configures the delay-start duration on a Cisco 3600 series voice port to 250 milliseconds:

voice-port 1/0/0
 timing delay-start 250
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing dial-pulse min-delay

To specify the time between wink-like pulses for a specified voice port on the Cisco 3600 series, use the timing dial-pulse min-delay voice-port configuration command. Use the no form of this command to reset the default value.

timing dial-pulse min-delay milliseconds
no timing dial-pulse min-delay milliseconds

Syntax Description

milliseconds

Time, in milliseconds, between the generation of wink-like pulses. Valid entries are numbers from 0 to 5000.

Default

300 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the timing command with the dial-pulse min-delay keyword with PBXes requiring a wink-like pulse, even though they have been configured for delay-dial signaling. If the value for this keyword is set to 0, the router will not generate this wink-like pulse. The call signal direction for this command is in.

Example

The following example configures the time between the generation of wink-like pulses on a Cisco 3600 series voice port to 350 milliseconds:

voice-port 1/0/0
 timing dial-pulse min-delay 350
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing dialout-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing digit

To specify the DTMF digit signal duration for a specified voice port, use the timing digit voice-port configuration command. Use the no form of this command to reset the default value.

timing digit milliseconds
no timing digit milliseconds

Syntax Description

milliseconds

The DTMF digit signal duration, in milliseconds. Valid entries are numbers from 50 to 100. Supported on FXO, FXS and E&M ports.

Default

100 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call signal direction for the timing digit command is out.

Examples

The following example configures the DTMF digit signal duration on a Cisco 3600 series voice port to 50 milliseconds:

voice-port 1/0/0
 timing digit 50
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing interdigit

To specify the DTMF interdigit duration for a specified voice port, use the timing interdigit voice-port configuration command. Use the no form of this command to reset the default value.

timing interdigit milliseconds
no timing interdigit milliseconds

Syntax Description

milliseconds

DTMF interdigit duration, in milliseconds. Valid entries are numbers from 50 to 500. Supported on FXO, FXS and E&M ports.

Default

100 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call signal direction for the timing interdigit command is out.

Examples

The following example configures the DTMF interdigit duration on a Cisco 3600 series voice port to 150 milliseconds:

voice-port 1/0/0
 timing interdigit 150
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing pulse
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing pulse

To specify the pulse dialing rate for a specified voice port, use the timing pulse voice-port configuration command. Use the no form of this command to reset the default value.

timing pulse pulses-per-second
no timing pulse pulses-per-second

Syntax Description

pulses-per-second

Pulse dialing rate, in pulses per second. Valid entries are numbers from 10 to 20. Supported on FXO and E&M ports only.

Default

20

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call signal direction for the timing pulse command is out.

Examples

The following example configures the pulse dialing rate on a Cisco 3600 series voice port to 15 pulses per second:

voice-port 1/0/0
 timing pulse 15
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse-interdigit
timing wink-duration
timing wink-wait

timing pulse-interdigit

To specify the pulse interdigit timing for a specified voice port, use the timing pulse-interdigit voice-port configuration command. Use the no form of this command to reset the default value.

timing pulse-interdigit milliseconds
no timing pulse-interdigit milliseconds

Syntax Description

milliseconds

Pulse dialing interdigit timing, in milliseconds. Valid entries are numbers from 100 to 1000. Supported on FXO and E&M ports only.

Default

500 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call signal direction for the timing pulse-interdigit command is out.

Examples

The following example configures the pulse-dialing interdigit timing on a Cisco 3600 series voice port to 300 milliseconds:

voice-port 1/0/0
 timing pulse-interdigit 300
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing wink-duration
timing wink-wait

timing wink-duration

To specify the maximum wink signal duration for a specified voice port, use the timing wink-duration voice-port configuration command. Use the no form of this command to restore the default value.

timing wink-duration milliseconds
no timing wink-duration milliseconds

Syntax Description

milliseconds

Maximum wink signal duration, in milliseconds, for a wink-start signal. Valid entries are numbers from 100 to 400. Supported on E&M ports only.

Default

200 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call signal direction for the timing wink-duration command is out.

Examples

The following example configures the wink signal duration on a Cisco 3600 series voice port to 300 milliseconds:

voice-port 1/0/0
 timing wink-duration 300
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-wait

timing wink-wait

To specify the maximum wink-wait duration for a specified voice port, use the timing wink-wait voice-port configuration command. Use the no form of this command to reset the default value.

timing wink-wait milliseconds
no timing wink-wait milliseconds

Syntax Description

milliseconds

Maximum wink-wait duration, in milliseconds, for a wink start signal. Valid entries are numbers from 100 to 5000. Supported on E&M ports only

Default

200 milliseconds

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

The call signal direction for the timing wink-wait command is out.

Examples

The following example configures the wink-wait duration on a Cisco 3600 series voice port to 300 milliseconds:

voice-port 1/0/0
 timing wink-wait 300
Related Commands

You can use the master indexes or search online to find documentation of related commands.

timeouts initial
timeouts interdigit
timing clear-wait
timing delay-duration
timing delay-start
timing dial-pulse min-delay
timing digit
timing interdigit
timing pulse
timing pulse-interdigit
timing wink-duration

type

To specify the E&M interface type, use the type voice-port configuration command. Use the no form of this command to reset the default value for this command.

type {1 | 2 | 3 | 5}
no type

Syntax Description

1

Indicates the following lead configuration:
E—output, relay to ground.
M—input, referenced to ground.

2

Indicates the following lead configuration:
E—output, relay to SG.
M—input, referenced to ground.
SB—feed for M, connected to -48V.
SG—return for E, galvanically isolated from ground.

3

Indicates the following lead configuration:
E—output, relay to ground.
M—input, referenced to ground.
SB—connected to -48V.
SG—connected to ground.

5

Indicates the following lead configuration:
E—output, relay to ground.
M—input, referenced to -48V.

Default

The default value is 1.

Command Mode

Voice-port configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the type command to specify the E&M interface for a particular voice port. With 1, the tie-line equipment generates the E-signal to the PBX type grounding the E-lead. The tie line equipment detects the M-signal by detecting current flow to ground. If you select 1, a common ground must exist between the line equipment and the PBX.

With 2, the interface requires no common ground between the equipment, thereby avoiding ground loop noise problems. The E-signal is generated toward the PBX by connecting it to SG. M-signal is indicated by the PBX connecting it to SB. While Type 2 interfaces do not require a common ground, they do have the tendency to inject noise into the audio paths because they are asymmetrical with respect to the current flow between devices.

With 3, the interface operates the same as Type 1 interfaces with respect to the E-signal. The M-signal, however, is indicated by the PBX connecting it to SB on assertion and alternately connecting it to SG during inactivity. If you select 3, a common ground must be shared between equipment.

With 5, the Type 5 line equipment indicates E-signal to the PBX by grounding the E-lead. The PBX indicates M-signal by grounding the M-lead. A Type 5 interface is quasi-symmetrical in that while the line is up, current flow is more or less equal between the PBX and the line equipment but noise injection is a problem.

Example

The following example selects Type 3 as the interface type for your voice port:

voice-port 1/0/0
 type 3
 
 
 

vad

To enable voice activity detection (VAD) for the calls using this dial peer, use the vad dial-peer configuration command. Use the no form of this command to disable this feature.

vad
no vad


Syntax Description

This command has no arguments or keywords.

Default

Enabled

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

Use the vad command to enable voice activity detection. With VAD, silence is not transmitted over the network, only audible speech. If you enable VAD, the sound quality will be slightly degraded but the connection will monopolize much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously transmitted to the IP backbone.

This command is applicable only to VoIP peers.

Example

The following example enables VAD:

dial-peer voice 10 voip
 vad
Related Commands

You can use the master index or search online to find documentation of related commands.

comfort-noise

voice-port

To enter the voice-port configuration mode, use the voice-port global configuration command.

voice-port slot-number/subunit-number/port

Syntax Description

slot-number/

Specifies the slot number in the Cisco router where the voice network module is installed. Valid entries are from 0 to 3, depending on the voice interface card you have installed.

subunit-number/

Specifies the subunit on the voice network module where the voice port is located. Valid entries are 0 or 1.

port

Specifies the voice port. Valid entries are 0 or 1.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3(1)T.

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

For more information about the physical characteristics of the voice network module, or how to install it, refer to installation documentation that came with your voice network module.

Example

The following example accesses the voice-port configuration mode for port 0, located on subunit 0 on a voice interface card installed in slot 1:

configure terminal
 voice-port 1/0/0
Related Commands

You can use the master index or search online to find documentation of related commands.

dial-peer