Voice-Related Commands

This chapter documents voice-related commands used to configure Voice over IP, Voice over Frame Relay, Voice over HDLC, and Voice over ATM. Commands in this section are listed alphabetically. For this release, Voice over IP is specific to the Cisco 3600 series; Voice over Frame Relay, Voice over ATM, Voice over HDLC, and Frame Relay-ATM Interworking are specific to the Cisco MC3810. For information on how to configure Voice over IP, Voice over Frame Relay, Voice over ATM, or Voice over HDLC, refer to the Voice, Video, and Home Applications Configuration Guide.

Table 3 describes which commands apply to each protocol.

Table 3: Voice-Related Commands Per Protocol
Commands Voice over IP Voice over Frame Relay Voice over ATM Voice over HDLC Frame Relay-ATM Interworking

acc-qos

X

alt-dial

X

X

X

answer-address

X

atm compression

X

cablelength long

X

X

X

cablelength short

X

X

X

channel-group

X

X

X

clear voice port

X

X

X

clock rate line

X

X

X

clock rate network-clock

X

X

X

clock source

X

X

X

codec

X

compress

X

cross-connect

X

X

X

X

destination-pattern

X

X

X

X

dial-control-mib

X

dial-peer terminator

X

dial-peer voice

X

X

X

X

X

encapsulation

X

X

expect-factor

X

fax-rate

X

forward-digits

X

X

X

frame-relay interface-dlci

X

frame-relay route

X

fr-atm connect dlci

X

icpif

X

interface fr-atm

X

ip precedence

X

ip udp checksum

X

loopback

X

loop-detect

X

X

X

mode

X

network-clock base-rate

X

X

X

X

network-clock-select

X

X

X

X

network-clock-switch

X

X

X

X

num-exp

X

port

X

X

X

X

preference

X

X

X

prefix

X

X

X

X

req-qos

X

session protocol

X

session target

X

X

X

X

show call active voice

X

show call history voice

X

show dial-peer voice

X

X

X

X

show dialplan incall number

X

show dialplan number

X

X

X

X

show network-clocks

X

X

X

show num-exp

X

show voice call

X

X

X

show voice dsp

X

X

X

show voice port

X

X

X

X

shutdown (dial-peer)

X

X

X

X

shutdown (DS 1 link)

X

X

X

snmp enable peer-trap poor-qov

X

X

X

X

snmp-server enable traps

X

X

X

X

tdm-group

X

X

X

vad

X

vbr-rt

X

voice-encap

X

voice-group

X

X

X

voice local-
bypass

X

X

X

**Table 3: Voice-Related Commands Per Protocol**
Commands	Voice over IP	Voice over Frame Relay	Voice over ATM	Voice over HDLC	Frame Relay-ATM Interworking
acc-qos	X
alt-dial		X	X	X
answer-address	X
atm compression			X
cablelength long		X	X	X
cablelength short		X	X	X
channel-group		X	X	X
clear voice port		X	X	X
clock rate line		X	X	X
clock rate network-clock		X	X	X
clock source		X	X	X
codec	X
compress			X
cross-connect		X	X	X	X
destination-pattern	X	X	X	X
dial-control-mib	X
dial-peer terminator	X
dial-peer voice	X	X	X	X	X
encapsulation			X		X
expect-factor	X
fax-rate	X
forward-digits		X	X	X
frame-relay interface-dlci		X
frame-relay route		X
fr-atm connect dlci					X
icpif	X
interface fr-atm					X
ip precedence	X
ip udp checksum	X
loopback			X
loop-detect		X	X	X
mode			X
network-clock base-rate		X	X	X	X
network-clock-select		X	X	X	X
network-clock-switch		X	X	X	X
num-exp	X
port	X	X	X	X
preference		X	X	X
prefix	X	X	X	X
req-qos	X
session protocol	X
session target	X	X	X	X
show call active voice	X
show call history voice	X
show dial-peer voice	X	X	X	X
show dialplan incall number	X
show dialplan number	X	X	X	X
show network-clocks		X	X	X
show num-exp	X
show voice call		X	X	X
show voice dsp		X	X	X
show voice port	X	X	X	X
shutdown (dial-peer)	X	X	X	X
shutdown (DS 1 link)		X	X	X
snmp enable peer-trap poor-qov	X	X	X	X
snmp-server enable traps	X	X	X	X
tdm-group		X	X	X
vad	X
vbr-rt			X
voice-encap				X
voice-group		X	X	X
voice local- bypass		X	X	X

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 restore the default value.

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. This is the default.
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 ensure 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

best-effort

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is only applicable to Cisco 3600 series VoIP peers.

Use the acc-qos dial-peer configuration 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.

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 indexes or search online to find documentation of related commands.

req-qos

alt-dial

To configure an alternate dial-out string for dial peers on the Cisco MC3810, use the alt-dial dial-peer configuration command. Use the no form of this command to delete the alternate dial-out string.

alt-dial string
no alt-dial string

Syntax Description


string	The alternate dial-out string.

Default

No alternate dial-out string is configured.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Cisco MC3810 POTS, VoFR, VoATM, and VoHDLC dial peers.

The alt-dial command is used for the on-net-to-off-net alternative dialing function. The string replaces the destination-pattern string for dialing out.

Example

The following example configures an alternate dial-out string of 9,5559871:

alt-dial 9,5559871

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 the configured telephone number.

answer-address [+]string[t]
no answer-address

Syntax Description


+	(Optional) Character indicating an E.164 standard number.
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. · Comma (,), which inserts a pause between digits. · Period (.), which matches any entered digit.
t	(Optional) Control character indicating that the answer-address value is a variable length dial-string.

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.

This command is applicable to both Cisco 3600 series VoIP and POTS dial peers.

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.

There are certain areas in the world (for example, in certain European countries) where valid telephone numbers can vary in length. Use the optional control character t to indicate that a particular answer-address value is a variable-length dial-string. In this case, the system will not match the dialed numbers until the interdigit timeout value has expired.

Note The Cisco IOS software does not check the validity of the E.164 telephone number; it accepts 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 indexes or search online to find documentation of related commands.

destination-pattern
port
prefix
timeouts interdigit

atm compression

To specify the software compression mode on the Cisco MC3810, use the atm compression interface configuration command. Use the no form of this command to remove the compression mode setting.

atm compression {per-packet | per-interface | per-vc}
no atm compression {per-packet | per-interface | per-vc}

Syntax Description


per-packet	Specifies packet-by-packet compression mode (no history). This is the default.
per-interface	Specifies one context per interface (with history).
per-vc	Specifies one context for every virtual circuit (with history).

Default

per-packet

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to ATM configuration on the Cisco MC3810.

Example

The following example configures per-packet ATM compression:

interface atm0
atm compression per-packet

cablelength long

To set a cable length longer than 655 feet for a DS1 (building CSU/DSU) link, use the cablelength long controller configuration command. Use the no form of this command to reset the default value.

cablelength long {gain26 | gain36} {-15db | -22.5db | -7.5db | 0db}
no cablelength long

Syntax Description


gain26	Specifies the decibel pulse gain at 26 for the receive side sensitivity. This is the default pulse gain.
gain36	Specifies the decibel pulse gain at 36 for the receive side sensitivity.
-15db	Specifies the decibel pulse rate for the transmit side attenuation at -15 decibels.
-22.5db	Specifies the decibel pulse rate for the transmit side attenuation at -22.5 decibels.
-7.5db	Specifies the decibel pulse rate for the transmit side attenuation at -7.5 decibels.
0db	Specifies the decibel pulse rate for the transmit side attenuation at 0 decibels. This is the default pulse rate.

Default

Gain26 and 0db.

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

On the Cisco MC3810, this command is supported on T1 controllers only.

Note On the Cisco MC3810, if you enter the cablelength long command on a DSX-1 interface, the command will be rejected because it is used for CSU interfaces only.

Example

The following example configures the cable length for controller T1 0 on a Cisco MC3810 to a decibel pulse gain of 36 and a decibel pulse rate of -22.5 decibels:

controller t1 0
cablelength long gain36 -22.5db

Related Commands

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

cablelength short

cablelength short

To set a cable length (line build-out) of 655 feet or shorter for a DSX-1 link, use the cablelength short controller configuration command. Use the no form of this command to reset the default value for this command.

cablelength short {133 | 266 | 399 | 533 | 655}
no cablelength short

Syntax Description


133	Specifies a cable length from 0 to 133 feet.
266	Specifies a cable length from 134 to 266 feet.
399	Specifies a cable length from 267 to 399 feet.
533	Specifies a cable length from 400 to 533 feet.
655	Specifies a cable length from 534 to 655 feet.

Default

No cable length is configured.

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

On the Cisco MC3810, this command is supported on T1 controllers only.

Note On the Cisco MC3810, if you enter the cablelength short command on a CSU interface, the command will be rejected because it is used for DSX-1 interfaces only.

Example

The following example configures the cable length to 133 on controller T1 0 on the Cisco MC3810:

controller t1 0
cablelength short 133

Related Commands

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

cablelength long

channel-group

To configure a list of timeslots for voice channels on the T1 controller, use the channel-group controller configuration command. Use the no form of this command to delete the channel group.

channel-group channel-no timeslots timeslot-list speed {56 | 64}
no channel-group channel-no timeslots timeslot-list speed {56 | 64}

Syntax Description


channel-no	Channel number to identify the channel group. The valid range is from 0 to 23.
timeslot-list	A list of timeslots that make up the channel group. The valid range is from 1 to 24.
speed {56 \| 64}	The speed of the underlying DS0s: 56 kbps, or 64 kbps. Note If you specify 56 kbps, the channel group is limited to 14 channels on the Cisco MC3810 MultiFlex Trunk (MFT). Because 56 is the default, you should specify 64 if you need more than 14 channels.

Default

56 kbps (for T1).
64 kbps (for E1).

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

Note Channel groups, CAS voice groups, and TDM groups all use group numbers. All group numbers configured for channel groups, CAS voice groups and TDM groups must be unique on the local Cisco MC3810 concentrator. For example, you cannot use the same group number for a channel group and for a TDM group.

Example

The following example configures a channel group on controller T1 0 on a Cisco MC3810:

controller T1 0
channel-group 10 timeslots10 64

clear voice port

To clear voice port calls in progress on the Cisco MC3810, use the clear voice port privileged EXEC command.

clear voice port [slot/port]

Syntax Description


slot/port	(Optional) The voice port slot number and port number. If you do not specify a voice port, all calls on all voice ports are cleared.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

If you do not specify a voice port, all calls on all voice ports are cleared. A confirmation prompt is displayed.

Example

The following example clears all calls on voice port 1/2 on the Cisco MC3810:

clear voice port 1/2

clock rate line

To configure the line clock rate for serial ports 0 or 1 in DTE mode on the Cisco MC3810, use the clock rate line interface configuration command. Use the no form of this command to cancel the clock rate line value.

clock rate line rate
no clock rate line rate

Syntax Description


rate	Network clock line rate in bits per second. The range is from 56 kbps to 2048 kbps. The value entered should be a multiple of 8,000 of the value set for the network-clock base-rate command. There is no default rate.

Default

No clock rate is set.

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command specifies the rate of the incoming clock so that the appropriate internal clock scaling can be performed.

To configure the clock rate for a serial port in DTE mode, use the clock rate network-clock command.

Example

The following configures the clock rate on serial 1 in DTE mode:

interface serial 1
clock rate line 2048

Related Commands

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

clock rate network-clock
network-clock base-rate

clock rate network-clock

To configure the network clock speed for serial ports 0 or 1 in DCE mode on the Cisco MC3810, use the clock rate network-clock interface configuration command. Use the no form of this command to cancel the network clock speed value.

clock rate network-clock rate
no clock rate network-clock rate

Syntax Description


rate	Network clock speed in bits per second. The range is from 56 kbps to 2048 kbps. The value entered should be a multiple of the value set for the network-clock base-rate command. There is no default rate.

Default

No clock rate is set.

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command uses a synchronized clock on the serial port. The use of this command allows the clock on the serial port to be synchronized with the clock source of controller T1 0.

To configure the clock rate for a serial port in DTE mode, use the clock rate line command.

Example

The following configures the clock rate on serial 1 in DCE mode:

interface serial 1
clock rate network-clock 2048

Related Commands

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

clock rate line
network-clock base-rate

clock source

To configure the clock source of a DS1 link on a Cisco MC3810, use the clock source controller configuration command.

clock source {line | internal | loop-timed}

Syntax Description


line	Specifies that the DS1 link uses the recovered clock. The line value is the default clock source used when the MFT is installed.
internal	Specifies that the DS1 link uses the internal clock. The internal value is the default clock source used when the DVM is installed.
loop-timed	Specifies that the T1/E1 controller will take the clock from the Rx (line) and use it for Tx. This setting decouples the controller clock from the system-wide clock set with the network-clock-select command. The loop-timed clock enables the digital voice module (DVM) to connect to a PBX and to connect the multiflex trunk (MFT) to a central office when both the PBX and the central office function as DCE clock sources. This situation assumes that the PBX also takes the clocking from the central office thereby synchronizing the clocks on the DVM and the MFT.

Default

Line (when the MFT is installed)
Internal (when the DVM is installed)

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.1.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

Example

The following example configures the clock source for the MFT to internal, and the clock source for the DVM for line on a Cisco MC3810:

controller T1 0
clock source internal
 
controller T1 1
clock source line

Note You cannot configure the clock source to the line setting for both T1/E1 controllers at the same time.

codec (dial-peer)

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.

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. This is the default CODEC.

Default

g729r8

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is only applicable to Cisco 3600 series VoIP peers.

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.

Note The CODEC value for the Cisco MC3810 is configured on the voice port.

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

compress

To configure whether compression is applied to the voice connection on the ATM permanent virtual circuit (PVC) for the Cisco MC3810, use the compress ATM virtual circuit configuration command. Use the no form of this command to turn off compression.

compress
no compress

Syntax Description

This command has no arguments or keywords.

Default

No compression

Command Mode

ATM virtual circuit configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 12.0.

This command applies to the ATM configuration on the Cisco MC3810.

Example

The following example configures ATM PVC 20 to support voice compression:

pvc 20
compress

cross-connect

To cross-connect two groups of digital signal level 0s (DS0s) from two controllers on the Cisco MC3810, or to cross-connect the Universal I/O (UIO) serial port for pass-through traffic to a trunk controller, use the cross-connect global configuration command. Use the no form of this command to remove the cross-connect function for the given controller.

For pass-through between two controllers, use the following commands:

cross-connect id controller-1 tdm-group-no-1 controller-2 tdm-group-no-2
no cross-connect id controller-1 tdm-group-no-1 controller-2 tdm-group-no-2

For pass-through traffic from a UIO serial port to a trunk controller, use the following commands:

cross-connect id interface-serial controller tdm-group-no
no cross-connect id interface-serial controller tdm-group-no

Note The UIO serial port is either serial port 0 or 1.

Syntax Description

id

Unique ID assigned to this cross-connection. The valid range is from 0 to 31.

controller-1

Type of the first controller (T1 0, T1 1, or E1)

tdm-group-no-1

TDM group number assigned to the first controller.

controller-2

Type of the second controller (T1, E1 0, or E1 1).

tdm-group-no-2

TDM group number assigned to the second controller.

id	Unique ID assigned to this cross-connection. The valid range is from 0 to 31.
controller-1	Type of the first controller (T1 0, T1 1, or E1)
tdm-group-no-1	TDM group number assigned to the first controller.
controller-2	Type of the second controller (T1, E1 0, or E1 1).
tdm-group-no-2	TDM group number assigned to the second controller.

For pass-through between two controllers:

For pass-through from a UIO serial port to a controller:

id

Unique ID assigned to this cross connection.

interface-serial

Number of the serial port, either 0 or 1.

controller

Type of the controller. Enter one of the following: T1 0, T1 1, E1 0, or
E1 1.

tdm-group-no

TDM group number assigned to the controller.

id	Unique ID assigned to this cross connection.
interface-serial	Number of the serial port, either 0 or 1.
controller	Type of the controller. Enter one of the following: T1 0, T1 1, E1 0, or E1 1.
tdm-group-no	TDM group number assigned to the controller.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, Voice over HDLC, and Frame Relay-ATM Interworking on the Cisco MC3810.

Example

The following example configures a pass-through cross-connect from serial port 0 to controller T1 1 on TDM group 20.

cross-connect 10 serial0 T1 1 20

Related Commands

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

tdm-group

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 the configured prefix or telephone number.

destination-pattern [+]string[t]
no destination-pattern

Syntax Description


+	(Optional) Character indicating an E.164 standard number. The plus sign (+) is not supported on the Cisco MC3810.
string	Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters: · The asterisk () and pound sign (#) that appear on standard touch-tone dial pads. On the Cisco 3600 only, these characters cannot be used as leading characters in a string (for example, 650). · Comma (,), which inserts a pause between digits. · Period (.), which matches any entered digit (this character is used as a wildcard). On the Cisco 3600, the period cannot be used as a leading character in a string (for example, .650).
t	Control character indicating that the destination-pattern value is a variable length dial-string.

Default

enabled with a null string

Command Mode

Dial-peer configuration

Usage Guidelines

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

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

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. When a router receives voice data, it compares the called number (the full E.164 telephone number) in the packet header with the number configured as the destination pattern for the voice-telephony peer. The router then strips out the left-justified numbers corresponding to the destination pattern. If you have configured a prefix, the prefix is appended to the front of the remaining numbers, creating a dial string, which the router then dials. If all numbers in the destination pattern are stripped-out, the user receives a dial tone.

There are certain areas in the world (for example, in certain European countries) where valid telephone numbers can vary in length. Use the optional control character t to indicate that a particular destination-pattern value is a variable-length dial-string. In this case, the system does not match the dialed numbers until the interdigit timeout value has expired.

Note The Cisco IOS software does not check the validity of the E.164 telephone number; it accepts 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 indexes or search online to find documentation of related commands.

answer-address
prefix
timeouts interdigit

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 prevents 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 prevents 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.

This command applies to Voice over IP on the Cisco 3600 series.

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 terminator

To designate a special character to be used as a terminator for variable length dialed numbers, use the dial-peer terminator global configuration command. Use the no form of this command to disable the designated terminating character.

dial-peer terminator character
no dial-peer terminator character

Syntax Description


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

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 12.0.

This command applies only to the Cisco 3600 series.

There are certain areas in the world (for example, in certain European countries) where valid telephone numbers can vary in length. When a dialed-number string has been identified as a variable length dialed-number, the system waits until the configured value for the timeouts interdigits command has expired before placing the call. To avoid waiting until the interdigit timeout value has expired, you can designate a special character as a terminator---meaning that when you dial that character, the system no longer waits for any additional digits and places the call. Use the dial-peer terminator global configuration command to designate a particular character as a terminator.

Example

The following example configures # as the special terminating character for variable-length dialed-numbers:

configure terminal
dial-peer terminator #

Related Commands

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

answer-address
destination-pattern
timeouts interdigit

dial-peer voice

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

For the Cisco 3600 series:

dial-peer voice number {voip | pots}

For the Cisco MC3810:

dial-peer voice tag-number {pots | voatm | vofr | vohdlc}

Syntax Description

For the Cisco 3600 series Voice over IP:

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.

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.

For the Cisco MC3810:

tag-number

Digit(s) defining a particular dial peer. Defines the dial peer and assigns the protocol type to it. Valid entries are from 1 to 10000.

pots

Indicates that this is a POTS peer using basic telephone service.

voatm

Indicates that this is a Voice over ATM peer using the real-time AAL5 voice encapsulation on the ATM backbone network.

vofr

Indicates that this is a Voice over Frame Relay peer using encapsulation on the Frame Relay backbone network.

vohdlc

Indicates that this is a Voice over HDLC peer using Cisco serial encapsulation (HDLC) for voice.

tag-number	Digit(s) defining a particular dial peer. Defines the dial peer and assigns the protocol type to it. Valid entries are from 1 to 10000.
pots	Indicates that this is a POTS peer using basic telephone service.
voatm	Indicates that this is a Voice over ATM peer using the real-time AAL5 voice encapsulation on the ATM backbone network.
vofr	Indicates that this is a Voice over Frame Relay peer using encapsulation on the Frame Relay backbone network.
vohdlc	Indicates that this is a Voice over HDLC peer using Cisco serial encapsulation (HDLC) for voice.

Command Mode

Global configuration

Usage Guidelines

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

This command applies to all voice applications on both the Cisco MC3810 and the Cisco 3600 series.

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 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 indexes or search online to find documentation of related commands.

voice-port

encapsulation

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

encapsulation aal-encap
no encapsulation aal-encap

Note This document only describes the relevant encapsulation settings for Voice over ATM and Frame Relay-Interworking on the Cisco MC3810. For the full syntax of the encapsulation command, refer to the Cisco IOS Wide Area Networking Command Reference. Note also that the Cisco MC3810 does not support SVCs or VC classes.

Syntax Description


aal-encap	ATM adaptation layer (AAL) and encapsulation type. When aal5mux is specified, a protocol is required. Possible values for aal-encap are as follows: aal5mux frame---For a MUX-type virtual circuit for Frame Relay-ATM Interworking on the Cisco MC3810. aal5mux voice---For a MUX-type virtual circuit for Voice over ATM on the Cisco MC3810. aal5snap---The only encapsulation supported for Inverse ARP. Logical Link Control/Subnetwork Access Protocol (LLC/SNAP) precedes the protocol datagram.

Default

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

Command Mode

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

VC-class configuration (for a VC class)

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 T.

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

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

If the encapsulation command is not explicitly configured on an ATM PVC or SVC, the VC inherits the following default configuration (listed in order of next highest precedence):

Configuration of the encapsulation command in a VC class assigned to the PVC or SVC itself.
Configuration of the encapsulation command in a VC class assigned to the PVC or SVC ATM subinterface.
Configuration of the encapsulation command in a VC class assigned to the PVC or SVC ATM main interface.
Global default: aal-encap = aal5snap.

Examples

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

pvc 20
encapsulation aal5mux voice

The following example configures a PVC to support encapsulation for Frame Relay-ATM Interworking:

pvc 21
encapsulation aal5mux frame

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.

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

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command applies to Cisco 3600 series VoIP peers.

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.

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.

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

voice

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is only applicable to Cisco 3600 series VoIP peers.

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 rate 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.

Example

The following example configures a transmission speed 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 indexes or search online to find documentation of related commands.

codec

forward-digits

To configure forward digits for voice calls on the Cisco MC3810, use the forward-digits dial-peer configuration command. Use the no form of this command to restore the default value.

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

Syntax Description


num-digit	The number of digits to be forwarded. If the number of digits is longer than the length of a destination phone number, the length of the destination number is used.
all	Forward all digits. If "all" is used, the full length of the destination pattern will be used.

Default

No digits are forwarded.

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies only to the Cisco MC3810.

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

Example

The following example configures forward digits for a POTS dial peer on a Cisco MC3810:

dial-peer voice 1 pots
  destination-pattern 8...
  forward-digits all

Related Commands

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

destination-pattern

frame-relay interface-dlci

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

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

frame-relay interface-dlci dlci [protocol ip ip-address] (for a BOOTP server only)
no frame-relay interface-dlci dlci [protocol ip ip-address]

Syntax Description


dlci	DLCI number to be used on the specified subinterface.
ietf \| cisco	(Optional) Encapsulation type: Internet Engineering Task Force (IETF) Frame Relay encapsulation or Cisco Frame Relay encapsulation.
protocol ip ip-address	(Optional) Indicates the IP address of the main interface of a new router or access server onto which a router configuration file is to be automatically installed over a Frame Relay network. Use this option only when this device will act as the BOOTP server for automatic installation over Frame Relay.
voice-encap size	(Supported on the Cisco MC3810 only.) Specifies that data segmentation will be used to support Voice over Frame Relay. The voice encapsulation size denotes the data segmentation size. The valid range is from 80 to 1600 bytes. For a list of recommended data segmentation sizes, see the "Usage Guidelines" section.

Default

No DLCI is assigned.

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 10.0.

This command is typically used for subinterfaces; however, it can also be used on main interfaces. Using the frame-relay interface-dlci command on main interfaces enables the use of routing protocols on interfaces that use Inverse ARP. The frame-relay interface-dlci command on a main interface is also valuable for assigning a specific class to a single PVC where special characteristics are desired. Subinterfaces are logical interfaces associated with a physical interface. You must specify the interface and subinterface before you can use this command to assign any DLCIs and any encapsulation or broadcast options. See the "Example" section for the sequence of commands.

This command is required for all point-to-point subinterfaces; it is also required for multipoint subinterfaces for which dynamic address resolution is enabled. It is not required for multipoint subinterfaces configured with static address mappings.

Use the protocol ip ip-address option only when this router or access server will act as the BOOTP server for autoinstallation over Frame Relay.

When configuring the voice-encap option on the Cisco MC3810, set the data segmentation size based on the port access rate. Table 4 lists recommended data segmentation sizes for different port access rates. Also, when the voice-encap option is configured on the Cisco MC3810, all priority queuing, custom queuing, and weighted fair queuing is disabled on the interface.

Table 4:

Port Access Rate Recommended Data Segmentation Size¹

64 kbps

80 bytes

128 kbps

160 bytes

256 kbps

320 bytes

512 kbps

640 bytes

1536 kbps (full T1)

1600 bytes

2048 kbps (full E1)

1600 bytes

¹The data segmentation size is based for back-to-back Frame Relay. If sending traffic through an IGX with standard Frame Relay, subtract 6 bytes from the recommended data segmentation size.

Recommended Data Segmentation Sizes

For more information about automatically installing router configuration files over a Frame Relay network, see the "Loading System Images and Microcode" chapter in the Configuration Fundamentals Configuration Guide.

Example

The following example assigns DLCI 100 to serial subinterface 5.17:

! Enter interface configuration and begin assignments on interface serial 5
interface serial 5
! Enter subinterface configuration by assigning subinterface 17
interface serial 5.17
! Now assign a DLCI number to subinterface 5.17
frame-relay interface-dlci 100

Related Commands

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

frame-relay class

frame-relay route

Use the frame-relay route interface configuration command to specify the static route for PVC switching. Use the no form of this command to remove a static route.

frame-relay route in-dlci out-interface out-dlci [voice-encap size]
no frame-relay route in-dlci out-interface out-dlci [voice-encap size]

Syntax Description


in-dlci	DLCI on which the packet is received on the interface.
out-interface	Interface that the router or access server uses to transmit the packet.
out-dlci	DLCI that the router or access server uses to transmit the packet over the specified out-interface.
voice encap size	(Supported on the Cisco MC3810 only.) Specifies that data segmentation will be used to support Voice over Frame Relay. Note that the voice encapsulation applies only to the input DLCI side. The valid range is from 8 to 1600.

Default

No static route is specified.

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 10.0.

When used with voice, the frame-relay route command is applied on both interfaces. If the voice-encap option is specified on one interface, then the incoming frames on that interface are defragmented before being routed to the other interface. The outgoing frames on that interface are then fragmented after being routed from the other interface, and before transmission out the interface.

Examples

The following example configures a static route that allows packets in DLCI 100 and transmits packets out over DLCI 200 on interface serial 2:

frame-relay route 100 interface Serial2 200

The following example illustrates the commands you enter for a complete configuration that includes two static routes for PVC switching between interface serial 1 and interface serial 2:

interface Serial1
no ip address
encapsulation frame-relay
keepalive 15
frame-relay lmi-type ansi
frame-relay intf-type dce
frame-relay route 100 interface Serial2 200
frame-relay route 101 interface Serial2 201
clockrate 2000000

fr-atm connect dlci

To map a Frame Relay DLCI to an ATM virtual circuit descriptor (VCD) for the FRF.5 Frame Relay-ATM interworking function on the Cisco MC3810, use the fr-atm connect dlci interface configuration command. The encapsulation type of the current interface must be Frame Relay or Frame Relay 1490 (IETF). Use the no form of this command to remove the DLCI to VCD map.

fr-atm connect dlci dlci atm-interface pvc [name | [vpi/] vci]
no fr-atm connect dlci dlci atm-interface pvc [name | [vpi/] vci]

Syntax Description


dlci	The Frame Relay DLCI number.
atm-interface	The ATM interface mapped to the DLCI. The ATM interface must be a serial interface with ATM encapsulation. On the Cisco MC3810, the interface must be ATM 0.
pvc	Specifies the ATM PVC.
name	The ATM PVC name.
vpi/	The ATM virtual path identifier (VPI).
vci	The ATM virtual channel identifier (VCI).

Default

No Frame Relay-ATM mapping is configured.

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Frame Relay-ATM Interworking on the Cisco MC3810.

Note The Cisco MC3810 provides only network interworking (FRF.5). The Cisco MC3810 can be used with service interworking (FRF.8), which is provided by the carrier's ATM network equipment.

Example

The following example configures a Frame Relay-ATM Interworking connection on FR-ATM interface 20, in which Frame Relay DLCI 100 is mapped to ATM VCI/VPI 100/200 for ATM interface 0.

interface fr-atm 20
fr-atm connect dlci 100 atm0 100/200

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.

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. The default is 30.

Default

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is applicable only to Cisco 3600 series VoIP peers.

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

Example

The following example disables the icpif command:

dial-peer voice 10 voip
icpif 0

interface fr-atm

To create a Frame Relay-ATM Interworking interface on the Cisco MC3810 and to enter Frame Relay-ATM Interworking configuration mode, use the interface fr-atm global configuration command. Use the no form of this command to delete the Frame Relay-ATM Interworking interface.

interface fr-atm number
no interface fr-atm number

Syntax Description


number	The Frame Relay-ATM Interworking interface number. Valid range is from 0 to 20.

Default

Frame Relay-ATM Interworking interface 20 is configured by default.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Frame Relay-ATM Interworking on the Cisco MC3810 only.

Use the interface fr-atm command to enter Frame Relay-ATM interworking interface configuration mode. When you enter this command for the first time, an interface number is created dynamically. You can configure up to 21 Frame Relay-ATM interworking interfaces.

Note The Cisco MC3810 provides only network interworking (FRF.5). The Cisco MC3810 can be used with service interworking (FRF.8), which is provided by the carrier's ATM network equipment.

Example

The following example configures Frame Relay-ATM Interworking interface number 20:

interface fr-atm 20

Related Command

fr-atm connect dlci

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.

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

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is applicable to Cisco 3600 series VoIP peers.

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. You should use this command if the IP link utilization is high and the QoS for voice packets needs to have a higher priority than other IP packets. You should also use the ip precedence command if RSVP is not enabled, and you want to give voice packets a higher priority over other IP data traffic.

Example

The following example sets the IP precedence to 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

disabled

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is applicable to Cisco 3600 series VoIP peers.

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.

Example

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

dial-peer voice 10 voip
ip udp checksum

loopback

To place a Cisco MC3810 ATM interface into loopback mode, use the loopback interface configuration command. Use the no form of this command to remove the loopback.

loopback [diagnostic | line]
no loopback [diagnostic | line]

Syntax Description


diagnostic	(Optional) Places the interface into internal loopback at the physical layer interface module (PLIM).
line	(Optional) Places the interface into external loopback at the line. Packets loop from the ATM interface back to the ATM network. This is the default.

Default

Line

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.0.

This command applies to Voice over ATM on the Cisco MC3810.

This command is useful for testing because it loops all packets from the ATM interface back to the ATM interface in addition to directing the packets to the network.

Example

The following example loops all packets back to the ATM interface on ATM interface 0:

interface atm0
loopback line

Related Commands

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

loop-detect

loop-detect

To enable loop detection for T1, use the loop-detect controller configuration command. Use the no form of the command to cancel the loop detect operation.

loop-detect
no loop-detect

Syntax Description

This command has no arguments or keywords.

Default

Loop detection is disabled.

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

Example

The following example configures loop detection for controller T1 0:

controller t1 0
loop-detect

Related Commands

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

loopback

mode

To set the mode of the T1/E1 controller and enter specific configuration commands for each mode type, use the mode controller configuration command. Use the no form of this command to restore the default mode of the controller.

mode {atm | cas}
no mode {atm | cas}

Syntax Description


atm	Sets the controller into ATM mode and creates an ATM interface (ATM 0) on the Cisco MC3810. When ATM mode is enabled, no channel groups, CAS groups, CCS groups, or clear channels are allowed because ATM occupies all the DS0s on the T1/E1 trunk. When you set the controller to ATM mode, the controller framing is automatically set to ESF for T1 or CRC4 for E1. The linecode is automatically set to B8ZS for T1 or HDBC for E1. When you remove ATM mode by entering the no mode atm command, ATM interface 0 is deleted. ATM mode is supported only on controller 0 (T1 or E1 0).
cas	Sets the controller into channel associated signaling (CAS) mode, which allows you to create channel groups, CAS groups, and clear channels (both data and CAS modes). CAS mode is supported on both controllers 0 and 1.

Default

No mode is configured.

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to the Cisco MC3810 with the digital voice module (DVM) installed.

When no mode is selected, channel groups and clear channels (data mode) can be created using the channel group and tdm-group commands, respectively.

On the Cisco MC3810, some DS0s are used exclusively for different signaling modes. The DS0 channels have the following limitations when mixing different applications (such as voice and data) on the same network trunk:

On E1 controllers, DS0 16 is used exclusively for either CAS or CCS, depending on which mode is configured.
On T1 controllers, DS0 24 is used exclusively for CCS.

Examples

The following example configures ATM mode on controller T1 0. This step is required for Voice over ATM:

controller T1 0
mode atm

The following example configures CAS mode on controller T1 1:

controller T1 1
mode cas

Related Commands

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

channel-group
tdm-group
voice-group

network-clock base-rate

To configure the network clock base rate for universal I/O serial ports 0 and 1 on the Cisco MC3810, use the network-clock base-rate global configuration command. Use the no form of this command to disable the current network clock base rate.

network-clock base-rate {56k | 64k}
no network-clock base-rate {56k | 64k}

Syntax Description


56k	Sets the network clock base rate to 56 kilobits per second (kbps).
64k	Sets the network clock base rate to 64kbps.

Default

56 kbps

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, Voice over HDLC, and Frame Relay-ATM Interworking on the Cisco MC3810.

Example

The following example sets the network clock base-rate to 64 kbps:

network-clock base-rate 64k

Related Commands

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

network-clock-select
network-clock-switch

network-clock-select

To use the network clock source to provide timing to the system backplane pulse code modulation (PCM) bus, use the network-clock-select global configuration command. Use the no form of this command to cancel the network clock selection.

network-clock-select priority [serial 0 | system | controller]
no network-clock-select priority [serial 0 | system | controller]

Syntax Description


priority	Specifies the priority of the clock source.Valid entries are from 1 to 4. You can configure up to four clock sources. The higher the number of the clock source, the higher the priority. For example, clock source 1 has higher priority than clock source 2. When the higher priority clock source fails, after the delay specified using the network-clock-switch command, the next higher priority clock source is selected.
serial 0	(Optional) Specifies serial interface 0 as the clock source.
system	(Optional) Specifies the system clock as the clock source.
controller	(Optional) Specifies which controllers is the clock source. You can specify either the trunk controller (T1/E1 0) or the digital voice module (T1/E1/ 1).

Default

No network clock source is specified.

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, Voice over HDLC, and Frame Relay-ATM Interworking on the Cisco MC3810.

Example

The following example sets the priority of four network clock sources. When the clock source with the highest priority (controller T1 0) fails, the Cisco MC3810 switches the clock source to the second highest priority (controller T1 1).

network-clock-select 1 T1 0
network-clock-select 2 T1 1
network-clock-select 3 serial 0
network-clock-select 4 System

Related Commands

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

network-clock-switch

To configure the switch delay time to the next priority network clock source when the current network clock source fails, use the network-clock-switch global configuration command. Use the no form of this command to cancel the network clock delay time selection.

network-clock-switch [switch-delay | never] [restore-delay | never]
no network-clock-switch delay

Syntax Description


switch-delay	The delay time before the next priority network clock source is used when the current network clock source fails. The range is from 0 to 99 seconds. The default is 10 seconds.
never	(Optional) Indicates no delay time before the current network clock source recovers.
restore-delay	(Optional) The delay time before the current network clock source recovers. The range is from 0 to 99 seconds.
never	(Optional) Indicates no delay time before the next priority network clock source is used when the current network clock source fails.

Default

10 seconds

Command Mode

Global configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, Voice over HDLC, and Frame Relay-ATM Interworking on the Cisco MC3810.

Example

The following command switches the network clock source after 20 seconds and sets the delay time before the current network clock source recovers to 20 seconds:

network-clock-switch 20 20

Related Commands

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

network-clock-select

num-exp

To define how to expand an extension number into a particular destination pattern, use the num-exp global configuration command. Use the no form of this command to cancel the configured number expansion.

num-exp extension-number expanded-number
no 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.

Default

No number expansion is defined.

Command Mode

Global configuration

Usage Guidelines

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

This command applies to Voice over IP on the Cisco 3600 series.

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 seven-digit numbers to numbers containing less than seven digits.

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

Examples

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

num-exp 65541 14085555541

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

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

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.

For the Cisco 3600 series:

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

For the Cisco MC3810:

port slot/port
no port

Syntax Description


slot	(Cisco MC3810) Slot number where the voice interface card is installed. Valid entries are 1 or 0.
slot-number	(Cisco 3600 series) Slot number in the 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	(Cisco 3600 series) Subunit on the voice interface card in the router where the voice port is located. Valid entries are 0 or 1.
port	(Cisco 3600 series) Voice port number. Valid entries are 0 or 1. (Cisco MC3810) Voice port number. Valid entries are the following: · Analog voice ports: 1-6 · Digital voice ports: T1: 1-24 E1: 1-15, and 16-31

Default

No port is configured.

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command is applicable only to POTS peers on both the Cisco 3600 series and the Cisco MC3810.

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.

Examples

The following example associates a Cisco 3600 series 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

The following example associates a Cisco MC3810 dial peer with voice port 1/1:

port 1/1

preference

To indicate the preference order of the dial peer on the Cisco MC3810, use the preference dial-peer configuration command. Setting the preference is useful in selecting the desired dial peer when multiple dial peers are matched for a dial string. Use the no form of this command to remove the preference value on the voice port.

preference value
no preference value

Syntax Description


value	An integer value from 0 to 10, where the lower the number, the higher the preference. The default value is 0 (highest preference).

Default

0 (highest preference)

Command Mode

Dial-peer configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Cisco MC3810 POTS, Voice over Frame Relay, Voice over ATM, and Voice over HDLC dial peers.

Note If POTS and voice-network peers are mixed in the same hunt group, the POTS dial peers must have priority over the voice-network peers.

Examples

The following example configures POTS dial peer 10 to a preference of 1 and VoFR dial peer 20 to a preference of 2:

dial-peer voice 10 pots
preference 1
 
dial-peer voice 20 vofr
preference 2

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

Null string

Command Mode

Dial-peer configuration

Usage Guidelines

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

This command applies to Cisco 3600 series and Cisco MC3810 POTS peers.

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.

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 indexes 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

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.

This command is applicable only to Cisco 3600 series VoIP peers.

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 issue 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.

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 indexes or search online to find documentation of related commands.

acc-qos

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

cisco

Command Mode

Dial-peer configuration

Usage Guidelines

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

For Cisco IOS Release 12.0, Cisco Session Protocol (cisco) is the only applicable session protocol. This command is applicable only to Cisco 3600 series 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 indexes 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.

For the Cisco 3600 series Voice over IP:

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

For the Cisco MC3810 Voice over Frame Relay:

session target interface{FR-ATM interface | Serial interface dlci}
no session target interface {FR-ATM interface | Serial interface dlci}

For Cisco MC3810 Voice over ATM:

session target interface ATM interface pvc [name | vpi/vci | vci]
no session target interface ATM interface pvc [name | vpi/vci | vci]

For Cisco MC3810 Voice over HDLC:

session target interface serial-port-number
no session target interface serial-port-number

Syntax Description

For the Cisco 3600 series Voice over IP:

ipv4:destination-address

IP address of the dial peer.

dns:host-name

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

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

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

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

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

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

loopback:rtp

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

loopback:compressed

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

loopback:uncompressed

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

For the Cisco MC3810 Voice over Frame Relay:

interface

Specifies the interface number.

FR-ATM

Specifies a logical Frame Relay-ATM interface on the Cisco MC3810. The valid range for the Frame Relay-ATM interface is 0-20.

Serial

Specifies a serial interface on the Cisco MC3810. The valid range for interface is 0 to 1.

dlci

Specifies the Frame Relay DLCI. The valid range is from 16 to 1007.

For the Cisco MC3810 Voice over ATM:

interface

Specifies the interface number.

ATM interface

Specifies the ATM interface number on the Cisco MC3810. The only valid number is 0.

name

The PVC name.

vpi/vci

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

vci

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

For the Cisco MC3810 Voice over HDLC:

interface

Specifies the interface number.

serial-port-number

Specifies the serial port number on the Cisco MC3810. The valid range is 0 to 1.

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.

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

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

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

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

Examples

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

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

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

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

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

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

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

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

The following example configures a session target for Voice over Frame Relay on a Cisco MC3810 with a session target on serial port1 and a DLCI of 200:

dial-peer voice 11 vofr
destination-pattern 13102221111
session target Serial1 200

The following example configures a session target for Voice over ATM on a Cisco MC3810. The session target is sent to ATM interface 0, and for a PVC with a VCI of 20.

dial-peer voice 12 voatm
destination-pattern 13102221111
session target atm0 pvc 20

The following example configures a session target on serial port 0 for Voice over HDLC on a Cisco MC3810:

dial-peer voice 13 vohdlc
destination-pattern 13102221111
session target serial0

Related Commands

You can use the master indexes 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 has no arguments or keywords.

Command Mode

Privileged EXEC

Usage Guidelines

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

This command applies to Voice over IP on the Cisco 3600 series.

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 two 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 two points in the connection. Each dial peer creates a call leg, as shown in Figure 2.

Figure 2: Call Legs Example

These two call legs are associated by the connection ID. The connection ID is global across the voice network, so that you can associate two call legs on one router with two 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:

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

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

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

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

Table 5 provides an alphabetical listing of the fields in this output and a description of each field.

Table 5: 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.

GapFillWithSilence

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

GapFillWithPrediction

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

GapFillWithInterpolation

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

GapFillWithRedundancy

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

HiWaterPlayoutDelay

High water mark Voice Playout FIFO Delay during this call.

Index

Dial peer identification number.

InfoActivity

Active information transfer activity state for this call.

InfoType

Information type for this call.

InSignalLevel

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

LogicalIfIndex

Index number of the logical interface for this call.

LoWaterPlayoutDelay

Low water mark Voice Playout FIFO Delay during the call.

NoiseLevel

Active noise level for the call.

OnTimeRvPlayout

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

OutSignalLevel

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

PeerAddress

Destination pattern associated with this peer.

PeerId

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

PeerIfIndex

Voice port index number for this peer.

PeerSubaddress

Subaddress to which this call is connected.

ReceiveBytes

Number of bytes received by the peer during this call.

ReceiveDelay

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

ReceivePackets

Number of packets received by this peer during this call.

RemoteIPAddress

Remote system IP address for the VoIP call.

RemoteUDPPort

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

RoundTripDelay

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

SelectedQoS

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

SessionProtocol

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

SessionTarget

Session target of the peer used for the call.

SetupTime

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

TransmitBytes

Number of bytes transmitted from this peer during the call.

TransmitPackets

Number of packets transmitted from this peer during the call.

TxDuration

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

VADEnable

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

VoiceTxDuration

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

Related Commands

You can use the master indexes 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. Valid entries 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.

This command applies to Voice over IP on the Cisco 3600 series.

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:

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

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

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

Table 6 provides an alphabetical listing of the fields in this output and a description of each field.

Table 6: Show Call History Voice Field Descriptions
Field Description

ACOMLevel

Average ACOM level for this 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.

CoderTypeRate

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

ConnectionID

Global call identifier for the gateway call.

ConnectTime

Time the call was connected.

DisconnectCause

Description explaining why the call was disconnected.

DisconnectText

Descriptive text explaining the disconnect reason.

DisconnectTime

Time the call was disconnected.

FaxDuration

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

GapFillWithSilence

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

GapFillWithPrediction

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

GapFillWithInterpolation

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

GapFillWithRedundancy

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

HiWaterPlayoutDelay

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

Index

Index number identifying the voice-peer for this call.

InfoType

Information type for this call.

LogicalIfIndex

Index of the logical voice port for this call.

LoWaterPlayoutDelay

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

NoiseLevel

Average noise level for this call.

OnTimeRvPlayout

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

PeerAddress

Destination pattern or number to which this call is connected.

PeerId

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

PeerIfIndex

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

PeerSubAddress

Subaddress to which this call is connected.

ReceiveBytes

Number of bytes received by the peer during this call.

ReceiveDelay

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

ReceivePackets

Number of packets received by this peer during the call.

RemoteIPAddress

Remote system IP address for the call.

RemoteUDPPort

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

RoundTripDelay

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

SelectedQoS

Selected RSVP quality of service for the call.

SessionProtocol

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

SessionTarget

Session target of the peer used for the call.

SetUpTime

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

TransmitBytes

Number of bytes transmitted by this peer during the call.

TransmitPackets

Number of packets transmitted by this peer during the call.

TxDuration

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

VADEnable

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

VoiceTxDuration

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

Related Commands

You can use the master indexes 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.

For the Cisco 3600 series:

show dial-peer voice [number]

For the Cisco MC3810:

show dial-peer voice [number] [summary]

Syntax Description

For the Cisco 3600 series:

number

(Optional) A specific dial peer. This option 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.

For the Cisco MC3810:

number

(Optional) A specific dial peer. This option displays configuration information for a single dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.

summary

(Optional) Displays a summary of all voice dial peers.

Command Mode

Privileged EXEC

Usage Guidelines

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

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

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 Displays

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

router# show dial-peer voice 1
VoiceEncapPeer1
        tag = 1, dest-pat = \Q+14085291000',
        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:

router# 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 7 explains the fields contained in both of these examples.

Table 7: Show Dial Peer Voice Field Descriptions
Field Description

Accepted Calls

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 for this field 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 measure for this field 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 indexes 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	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	Subunit on the voice network module where the voice port is located. Valid entries are 0 or 1.
port	Voice port. Valid entries are 0 or 1.
dial string	Particular destination pattern (telephone number).

Command Mode

Privileged EXEC

Usage Guidelines

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

This command applies to Voice over IP on the Cisco 3600 series.

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 indexes 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 dialplan number privileged EXEC command.

show dialplan number dial string

Syntax Description


dial string	Particular destination pattern (telephone number).

Command Mode

Privileged EXEC

Usage Guidelines

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

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

The show dialplan number command is used to test if the dial-plan configuration is valid and working as expected.

Sample Display

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

router# show dialplan number 51234
 
Macro Exp.: 14085551234
VoiceOverIpPeer1004
        tag = 1004, destination-pattern = \Q+1408555....',
        answer-address = \Q',
        group = 1004, Admin state is up, Operation state is up
        type = voip, session-target = \Qipv4:1.13.24.0',
        ip precedence: 0        UDP checksum = disabled
        session-protocol = cisco, req-qos = best-effort,
        acc-qos = best-effort,
        fax-rate = voice, codec = g729r8,
        Expect factor = 10, Icpif = 30,
        VAD = enabled, 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
Matched: +14085551234   Digits: 7
Target: ipv4:172.13.24.0

Table 8 explains the fields contained in this example.

Table 8: Show DialPlan Number Field Descriptions
Field Description

Macro Exp.

Expected destination pattern for this dial peer.

VoiceOverIpPeer

Identifies the dial peer associated with the destination pattern entered.

tag

Unique dial peer identifying number

destination-pattern

Destination pattern (telephone number) configured for this dial peer

answer-address

Answer address configured for this dial peer.

Admin state

Describes the administrative state of this dial peer.

Operation state

Describes the operational state of the dial peer.

type

Type of dial peer (POTS or VoIP).

session-target

Displays the configures session target (IP address or host name) for this dial peer.

ip precedence

Displays the numeric value for the IP Precedence configured for this dial peer.

UDP checksum

Indicates the status of the UDP checksum feature.

session-protocol

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

req-qos

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

acc-qos

Configures acceptable quality of service for calls for this dial peer.

fax-rate

Configured facsimile transmission speed for with this dial peer.

codec

CODEC type configured for this dial peer.

Expect factor

Configured value at which the system will generate an SMTP message alerting that the voice quality has dropped.

Icpif

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

VAD

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

Poor QOV Trap

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

Connect Time

Unit of measure indicating the call connection time associated with this dial peer.

Charged Units

Number of call units charged to this dial peer.

Successful Calls

Number of completed calls to this peer since system startup.

Failed Calls

Number of uncompleted (failed) calls to this peer since system startup.

Accepted Calls

Number of calls from this peer accepted since system startup.

Refused Calls

Number of calls from this peer refused since system startup.

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.

Matched

Destination pattern matched for this dial peer.

Target

Matched session target (IP address or host name) for this dial peer.

Related Commands

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

show dialplan incall number

show network-clocks

To display the current configured and active network clock sources, use the show network-clocks privileged EXEC command.

show network-clocks

Syntax Description

This command has n arguments of keywords.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

The Cisco MC3810 has a background task that verifies whether a valid clocking configuration exists every 120 seconds. If this task detects an error, you will be reminded every 120 seconds until the error is corrected. A clocking configuration error may be generated for various reasons. Using the show network-clocks command, you can display the clocking configuration status.

Sample Displays

The following is sample output from the show network-clocks command on the Cisco MC3810:

router# show network-clocks
Priority 1 clock source(inactive config): T1 0
Priority 1 clock source(active config) : T1 0
Clock switch delay: 10
Clock restore delay: 10
T1 0 is clocking system bus for 9319 seconds.
Run Priority Queue: controller0

In this display, inactive configuration is the new configuration that has been established. Active configuration is the run-time configuration. Should an error be made in the new configuration, the inactive and active configurations will be different. In the above example, the clock priority configuration is valid, and the system is being clocked as indicated.

The following is another sample output from the show network-clocks command:

router# show network-clocks
Priority 1 clock source(inactive config) : T1 0
Priority 2 clock source(inactive config) : T1 1
Priority 1 clock source(active config) : T1 0
Clock switch delay: 10
Clock restore delay: 10
T1 0 is clocking system bus for 9319 seconds.
Run Priority Queue: controller0

In this display, the new clocking configuration has an error for controller T1 1. This is indicated by checking differences between the last valid configuration (active) and the new proposed configuration (inactive). The error may result from hardware (the system controller board or MFT) unable to support this mode, or controller T1 1 is currently configured as "clock source internal."

Since the active and inactive configurations are different, the system will periodically display the warning message about the wrong configuration.

show num-exp

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

show num-exp [dialed-number]

Syntax Description


dialed-number	(Optional) Dialed number.

Command Mode

Privileged EXEC

Usage Guidelines

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

This command applies to Voice over IP on the Cisco 3600 series.

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:

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

Table 9 explains the fields in the sample output.

Table 9: Show Num-Exp Voice Field Descriptions

Field Description

Dest Digit Pattern

Index number identifying the destination telephone number digit pattern.

Translation

Expanded destination telephone number digit pattern.

Related Commands

You can use the master indexes 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 call

To show the call status for all voice ports on the Cisco MC3810, use the show voice call privileged EXEC command.

show voice call [summary]

Syntax Description


summary	(Optional) Specifies to show a summary of the status instead of the full detailed report.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

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

Tandem switch
End-to-end call manager
Call processing state machine
Protocol state machine

Sample Display

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

router# show voice call summary
1/1 (orig): eecm = ST_DIGIT_COLLECT,  LFXS= call_progress,  CPD= failure_cont
 
1/2 (  ): eecm = IDLE,  LFXS= idle,  CPD= idle
 
1/3 (  ): eecm = IDLE,  LFXS= idle,  CPD= idle
 
1/4 (  ): eecm = IDLE,  LFXO= idle,  CPD= idle
 
1/5 (  ): eecm = IDLE,  LEM= idle,  CPD= idle
 
1/6 (  ): eecm = IDLE,  LEM= idle,  CPD= idle

Table 10 explains the fields in the sample output.

Table 10: Show Voice Call Field Descriptions

Field Description

(orig)

Indicates the call is originating on the voice port.

eecm

Status of the End-to-End Call Manager.

LFXS

Status of the FXS line.

CPD

Status of the Call Processing Data.

LFXO

Status of the FXO line.

LEM

Status of the E&M line.

Related Commands

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

show dial-peer voice
show voice dsp
show voice-port

show voice dsp

To show the current status of all DSP voice channels, use the show voice dsp privileged EXEC command.

show voice dsp

Syntax Description

This command has no arguments or keywords.

Command Mode

Privileged EXEC

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

Sample Display

The following is sample output from the show voice dsp command for the Cisco MC3810:

router# show voice dsp
DSP# 0, channel# 0 G729A BUSY
DSP# 0, channel# 1 G729A BUSY
DSP# 1, channel# 2 FAX IDLE
DSP# 1, channel# 3 FAX IDLE
DSP# 2, channel# 4 NONE BAD
DSP# 2, channel# 5 NONE BAD
DSP# 3, channel# 6 NONE BAD
DSP# 3, channel# 7 NONE BAD
DSP# 4, channel# 8 NONE BAD
DSP# 4, channel# 9 NONE BAD
DSP# 5, channel# 10 NONE BAD
DSP# 5, channel# 11 NONE BAD

Table 11 explains the fields in the sample output.

Table 11: Show Voice DSP Field Descriptions
Field Description

DSP

Number of the DSP

Channel

Number of the channel and its status.

Related Commands

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

show dial-peer voice
show voice call summary
show voice-port

show voice port

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

For the Cisco 3600 series:

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

For the Cisco MC3810:

show voice port [slot/port] [summary]

Syntax Description

For the Cisco 3600 series:

slot-number

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

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

port

Voice port number. Valid entries are 0 or 1.

For the Cisco MC3810:

slot/port

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

slot specifies the slot number in the Cisco router where the voice interface card is installed. The only valid entry is 1.

port specifies the voice port number. Valid ranges are as follows:

Analog voice ports: from 1 to 6.

Digital voice port:

Digital T1: from 1 to 24.

Digital E1: from 1 to 15, and from 17 to 31.

summary

(Optional) Display a summary of all voice ports.

Command Mode

Privileged EXEC

Usage Guidelines

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

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

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 on the Cisco 3600 series:

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

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

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

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

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

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

router# show voice port summary
 
                                                              IN  OUT   ECHO
PORT SIG-TYPE     ADMIN OPER IN-STATUS OUT-STATUS CODEC  VAD GAIN ATTN CANCEL
1/1  fxs-ls       up    up   on-hook    idle      729a    n    0    0    y
1/2  fxs-ls       up    up   on-hook    idle      729a    n    0    0    y
1/3  e&m-wnk      up    up   idle       idle      729a    n    0    0    y
1/4  e&m-wnk      up    up   idle       idle      729a    n    0    0    y
1/5  fxo-ls       up    up   idle       on-hook   729a    n    0    0    y
1/6  fxo-ls       up    up   idle       on-hook   729a    n    0    0    y

Table 12 explains the fields in the sample output.

Table 12: Show Voice Port Field Descriptions
Field Description

Administrative State

Administrative state of the voice port.

Alias

User-supplied alias for this voice port.

Analog interface A-D gain offset

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

Analog interface D-A gain offset

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

Clear Wait Duration Timing

Time of inactive seizure signal to declare call cleared.

Coder Type

Voice compression mode used.

Companding Type

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

Connection Mode

Connection mode of the interface.

Connection Number

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

Currently Processing

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

Delay Duration Timing

Maximum delay signal duration for delay dial signaling.

Delay Start Timing

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

Description

Description of the voice port.

Dial Type

Out-dialing type of the voice port.

Digit Duration Timing

DTMF Digit duration in milliseconds.

E&M Type

Type of E&M interface.

Echo Cancel Coverage

Echo Cancel Coverage for this port.

Echo Cancellation

Whether or not echo cancellation is enabled for this port.

Hook Flash Duration Timing

Maximum length of hook flash signal.

Hook Status

Hook status of the FXO/FXS interface.

Impedance

Configured terminating impedance for the E&M interface.

In Gain

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

In Seizure

Incoming seizure state of the E&M interface.

Initial Time Out

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

InterDigit Duration Timing

DTMF interdigit duration in milliseconds.

InterDigit Pulse Duration Timing

Pulse dialing interdigit timing in milliseconds.

Interdigit Time Out

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

Maintenance Mode

Maintenance mode of the voice port.

Maximum Playout Delay

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

Music On Hold Threshold

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

Noise Regeneration

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

Nominal Playout Delay

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

Non-Linear Processing

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

Number of signaling protocol errors

Number of signaling protocol errors.

Operations State

Operation state of the port.

Operation Type

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

Out Attenuation

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

Out Seizure

Outgoing seizure state of the E&M interface.

Port

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

Pulse Rate Timing

Pulse dialing rate in pulses per second (pps).

Region Tone

Configured regional tone for this interface.

Ring Active Status

Ring active indication.

Ring Cadence

Configured ring cadence for this interface.

Ring Frequency

Configured ring frequency for this interface.

Ring Ground Status

Ring ground indication.

Ringing Time Out

Ringing time out duration.

Signal Type

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

Slot

Slot used in the voice interface card for this port.

Sub-unit

Subunit used in the voice interface card for this port.

Tip Ground Status

Tip ground indication.

Type of VoicePort

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

The Interface Down Failure Cause

Text string describing why the interface is down,

Voice Activity Detection

Whether Voice Activity Detection is enabled or disabled.

Wait Release Time Out

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

Wink Duration Timing

Maximum wink duration for wink start signaling.

Wink Wait Duration Timing

Maximum wink wait duration for wink start signaling.

Related Commands

You can use the master indexes 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)

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.

Default

no shutdown

Command Mode

Dial-peer configuration

Usage Guidelines

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

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

When a dial peer is shut down, you cannot initiate calls to that peer. This command is applicable to both Cisco 3600 series and Cisco MC3810 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 (DS1 Link)

To shutdown a DS1 link (send a Blue Alarm), use the shutdown controller configuration command. Use the no form of the command to activate the DS1 (cancel the sending of the Blue Alarm).

shutdown
no shutdown

Syntax Description

This command has no arguments or keywords.

Default

no shutdown

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

Example

The following example shuts down a DS 1 link on controller T1 0:

controller T1 0
shutdown

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 notification.

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.

This command is applicable only to VoIP peers on the Cisco 3600 series; VoFR, VoATM, VoHDLC peers on the Cisco MC3810.

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.

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 indexes 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. · reset---Enables 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 applies to Voice over IP on the Cisco 3600 series, Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

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 issue 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

tdm-group

To configure a list of time slots for creating clear channel groups (pass-through) for Time Division Multiplexing (TDM) cross-connect, use the tdm-group controller configuration command. Use the no form of this command to delete a clear channel group.

tdm-group tdm-group-no timeslot timeslot-list [type {e&m | fxs [loop-start | ground-start] | fxo [loop-start | ground-start] | fxs-melcas | fxo-melcas | e&m-melcas}]
no tdm-group tdm-group-no

Syntax Description


tdm-group-no	Time Division Multiplexing (TDM) group number.
timeslot	Timeslot number.
timeslot-list	Timeslot list. The valid range is from 1-24 for T1, and from 1-15 and 17-31 for E1.
type	(Optional) (Valid only when the mode cas command is enabled.) Specifies the voice signaling type of the voice port. If configuring a TDM group for data traffic only, do not specify the type option. Choose from one of the following options:
	· e&m---for E&M signaling · fxo---for Foreign Exchange Office signaling (optionally, you can also specify loop-start or ground-start) · fxs---for Foreign Exchange Station signaling (optionally, you can also specify loop-start or ground-start) · e&m-melcas---for E&M Mercury Exchange Limited (MEL) Channel Associated Signaling · fxs-melcas--- for Foreign Exchange Station Mercury Exchange Limited (MEL) Channel Associated Signaling · fxo-melcas---for Foreign Exchange Office Mercury Exchange Limited (MEL) Channel Associated Signaling The melcas options apply only to E1 lines and are used primarily in the United Kingdom.

Default

No TDM group is configured.

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

Example

The following example configures TDM group number 20 on controller T1 1 to support FXO ground-start:

controller T1 1
mode cas
tdm-group 20 20 type fxs ground-start

Related Commands

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

mode

vad (dial peer)

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 VAD.

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.

This command applies to Voice over IP on the Cisco 3600 series.

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 is slightly degraded but the connection monopolizes 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.

Note On the Cisco MC3810, VAD is assigned to the voice port using the vad voice-port configuration command.

Example

The following example enables VAD:

dial-peer voice 10 voip
vad

Related Commands

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

comfort-noise

vbr-rt

To configure the real-time variable bit rate (VBR) for voice connections on the Cisco MC3810, use the vbr-rt ATM virtual circuit configuration command. Use the no form of this command to restore the default value.

vbr-rt peak-rate average-rate burst
no vbr-rt peak-rate average-rate burst

Syntax Description


peak-rate	The peak information rate (PIR) of the voice connection in kbps. The range is from 56 to 10,000.
average-rate	The average information rate (AIR) of the voice connection in kbps. The range is from 1 to 56.
burst	Burst size in number of cells. The range is from 0 to 65536.

Default

No vbr-rt settings are configured.

Command Mode

ATM virtual circuit configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 12.0.

This command applies to Voice over ATM on the Cisco MC3810.

The vbr-rt command configures traffic shaping between voice and data PVCs. Traffic shaping is required so that the carrier does not discard calls. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will be carrying bursty traffic. The peak, average, and burst values are needed so the PVC can effectively handle the bandwidth for the number of voice calls. To calculate the minimum peak, average, and burst values for the number of voice calls, use the following calculations:

Peak value: (2 x the maximum number of calls) x 16 kb
Average value: (1 x the maximum number of calls) x 16 kb
Burst value: (4 x the maximum number of calls)

Note When you configure data PVCs that will be traffic shaped with voice PVCs, use the aalsnap encapsulation and calculate the overhead as 1.13 times the voice rate.

Example

The following example configures the traffic shaping rate for ATM PVC 20 on a Cisco MC3810. In the example, the peak, average and burst rates are calculated based on a maximum of 20 calls on the PVC.

pvc 20
encapsulation aal5mux voice
vbr-rt 640 320 80

Related Commands

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

encapsulation

voice-encap

To define the data segmentation size on an HDLC interface to support voice over HDLC, use the voice-encap interface configuration command. Use the no form of this command to delete the setting.

voice-encap size
no voice-encap size

Syntax Description


size	The size of the data segmentation. The valid range is from 8 to 1600.

Default

No data segmentation size is defined.

Command Mode

Interface configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over HDLC on the Cisco MC3810.

Example

The following example configures serial interface 1 to support a data segmentation size of 64 for Voice over HDLC:

interface serial 1
voice-encap 64

voice-group

To configure a list of timeslots for voice channel associated signaling (CAS) on the T1/E1 controller on the Cisco MC3810, use the voice-group controller configuration command. Use the no form of this command to delete the CAS group.

voice-group voice-group-no timeslots timeslot-list type {e&m-immediate | e&m-delay |
e&m-wink | e&m-melcas | fxs-ground-start | fxs-loop-start | fxs-melcas |
fxo-ground-start | fxo-loop-start | fxo-melcas}
no voice-group voice-group-no timeslots timeslot-list

Syntax Description


voice-group-no	Number to identify the voice group. The valid range is from 0 to 23.
timeslot timeslot-list	A list of timeslots that makes up the CAS group. The valid range is from 1 to 24 for T1, and from 1 to 15 and 17 to 31 for E1.
type	The type of voice interface: · e&m-immediate---for E&M immediate · e&m-delay---for E&M delay · e&m-wink---for E&M wink · e&m-melcas---for E&M Mercury Exchange Limited (MEL) Channel Associated Signaling (equivalent to CEPT) · fxs-ground-start---for Foreign Exchange Station ground-start · fxs-loop-start---for Foreign Exchange Station loop-start · fxs-melcas--- for Foreign Exchange Station Mercury Exchange Limited (MEL) Channel Associated Signaling · fxo-ground-start---for Foreign Exchange Office.ground-start · fxo-loop-start---for Foreign Exchange Office.loop-start · fxo-melcas---for Foreign Exchange Office Mercury Exchange Limited (MEL) Channel Associated Signaling The melcas options are used primarily in the United Kingdom.

Default

No voice group is configured.

Command Mode

Controller configuration

Usage Guidelines

This command first appeared in Cisco IOS Release 11.3 MA.

This command applies to Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810.

This command is only available if the mode cas command is enabled.

Example

The following example configures a voice group on controller T1 0 on a Cisco MC3810:

controller T1 0
mode cas
voice-group 10 timeslots10 64

This command allows you to pass uncompressed voice traffic for local POTS calls.

Example

The following example configures the Cisco MC3810 to directly cross-connect local calls without going through a Digital Signal Processor (DSP):

voice local-bypass

Port Access Rate	Recommended Data Segmentation Size¹
64 kbps	80 bytes
128 kbps	160 bytes
256 kbps	320 bytes
512 kbps	640 bytes
1536 kbps (full T1)	1600 bytes
2048 kbps (full E1)	1600 bytes

ipv4:destination-address	IP address of the dial peer.
dns:host-name	Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. (Optional) You can use one of the following three wildcards with this keyword when defining the session target for VoIP peers: · $s$.---Indicates that the source destination pattern will be used as part of the domain name. · $d$.---Indicates that the destination number will be used as part of the domain name. · $e$.---Indicates that the digits in the called number will be reversed, periods will be added in-between each digit of the called number, and that this string will be used as part of the domain name. · $u$.---Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name.
loopback:rtp	Indicates that all voice data will be looped back to the originating source. This is applicable for VoIP peers.
loopback:compressed	Indicates that all voice data will be looped back in compressed mode to the originating source. This is applicable for POTS peers.
loopback:uncompressed	Indicates that all voice data will be looped-back in uncompressed mode to the originating source. This is applicable for POTS peers.

interface	Specifies the interface number.
FR-ATM	Specifies a logical Frame Relay-ATM interface on the Cisco MC3810. The valid range for the Frame Relay-ATM interface is 0-20.
Serial	Specifies a serial interface on the Cisco MC3810. The valid range for interface is 0 to 1.
dlci	Specifies the Frame Relay DLCI. The valid range is from 16 to 1007.

interface	Specifies the interface number.
ATM interface	Specifies the ATM interface number on the Cisco MC3810. The only valid number is 0.
name	The PVC name.
vpi/vci	The ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC.
vci	The ATM network virtual channel identifier (VCI) of this PVC.

interface	Specifies the interface number.
serial-port-number	Specifies the serial port number on the Cisco MC3810. The valid range is 0 to 1.

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.
GapFillWithSilence	Duration of voice signal replaced with silence because voice data was lost or not received on time for this call.
GapFillWithPrediction	Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. An example of such pullout is frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms.
GapFillWithInterpolation	Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received on time from voice gateway for this call.
GapFillWithRedundancy	Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received on time from voice gateway for this call.
HiWaterPlayoutDelay	High water mark Voice Playout FIFO Delay during this call.
Index	Dial peer identification number.
InfoActivity	Active information transfer activity state for this call.
InfoType	Information type for this call.
InSignalLevel	Active input signal level from the telephony interface used by this call.
LogicalIfIndex	Index number of the logical interface for this call.
LoWaterPlayoutDelay	Low water mark Voice Playout FIFO Delay during the call.
NoiseLevel	Active noise level for the call.
OnTimeRvPlayout	Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
OutSignalLevel	Active output signal level to telephony interface used by this call.
PeerAddress	Destination pattern associated with this peer.
PeerId	ID value of the peer table entry to which this call was made.
PeerIfIndex	Voice port index number for this peer.
PeerSubaddress	Subaddress to which this call is connected.
ReceiveBytes	Number of bytes received by the peer during this call.
ReceiveDelay	Average Playout FIFO Delay plus the decoder delay during the voice call.
ReceivePackets	Number of packets received by this peer during this call.
RemoteIPAddress	Remote system IP address for the VoIP call.
RemoteUDPPort	Remote system UDP listener port to which voice packets are transmitted.
RoundTripDelay	Voice packet round trip delay between the local and remote system on the IP backbone during the call.
SelectedQoS	Selected RSVP quality of service (QoS) for the call.
SessionProtocol	Session protocol used for an Internet call between the local and remote router via the IP backbone.
SessionTarget	Session target of the peer used for the call.
SetupTime	Value of the System UpTime when the call associated with this entry was started.
TransmitBytes	Number of bytes transmitted from this peer during the call.
TransmitPackets	Number of packets transmitted from this peer during the call.
TxDuration	Duration of transmit path open from this peer to the voice gateway for the call.
VADEnable	Whether or not voice activation detection (VAD) was enabled for this call.
VoiceTxDuration	Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value.

Field	Description
ACOMLevel	Average ACOM level for this 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.
CoderTypeRate	Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for the call.
ConnectionID	Global call identifier for the gateway call.
ConnectTime	Time the call was connected.
DisconnectCause	Description explaining why the call was disconnected.
DisconnectText	Descriptive text explaining the disconnect reason.
DisconnectTime	Time the call was disconnected.
FaxDuration	Duration of fax transmitted from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing this value by the TxDuration value.
GapFillWithSilence	Duration of voice signal replaced with silence because the voice data was lost or not received on time for this call.
GapFillWithPrediction	Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call.
GapFillWithInterpolation	Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call.
GapFillWithRedundancy	Duration of voice signal played out with signal synthesized from redundancy parameters available because the voice data was lost or not received on time from the voice gateway for this call.
HiWaterPlayoutDelay	High water mark Voice Playout FIFO Delay during the voice call.
Index	Index number identifying the voice-peer for this call.
InfoType	Information type for this call.
LogicalIfIndex	Index of the logical voice port for this call.
LoWaterPlayoutDelay	Low water mark Voice Playout FIFO Delay during the voice call.
NoiseLevel	Average noise level for this call.
OnTimeRvPlayout	Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values.
PeerAddress	Destination pattern or number to which this call is connected.
PeerId	ID value of the peer entry table to which this call was made.
PeerIfIndex	Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for the call.
PeerSubAddress	Subaddress to which this call is connected.
ReceiveBytes	Number of bytes received by the peer during this call.
ReceiveDelay	Average Playout FIFO Delay plus the decoder delay during the voice call.
ReceivePackets	Number of packets received by this peer during the call.
RemoteIPAddress	Remote system IP address for the call.
RemoteUDPPort	Remote system UDP listener port to which voice packets for this call are transmitted.
RoundTripDelay	Voice packet round trip delay between the local and remote system on the IP backbone for this call.
SelectedQoS	Selected RSVP quality of service for the call.
SessionProtocol	Session protocol to be used for an Internet call between the local and remote router via the IP backbone.
SessionTarget	Session target of the peer used for the call.
SetUpTime	Value of the System UpTime when the call associated with this entry was started.
TransmitBytes	Number of bytes transmitted by this peer during the call.
TransmitPackets	Number of packets transmitted by this peer during the call.
TxDuration	Duration of the transmit path open from this peer to the voice gateway for the call.
VADEnable	Whether or not voice activation detection (VAD) was enabled for this call.
VoiceTxDuration	Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value.

number	(Optional) A specific dial peer. This option displays configuration information for a single dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767.
summary	(Optional) Displays a summary of all voice dial peers.

Field	Description
Accepted Calls	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 for this field 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 measure for this field 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.

Field	Description
Macro Exp.	Expected destination pattern for this dial peer.
VoiceOverIpPeer	Identifies the dial peer associated with the destination pattern entered.
tag	Unique dial peer identifying number
destination-pattern	Destination pattern (telephone number) configured for this dial peer
answer-address	Answer address configured for this dial peer.
Admin state	Describes the administrative state of this dial peer.
Operation state	Describes the operational state of the dial peer.
type	Type of dial peer (POTS or VoIP).
session-target	Displays the configures session target (IP address or host name) for this dial peer.
ip precedence	Displays the numeric value for the IP Precedence configured for this dial peer.
UDP checksum	Indicates the status of the UDP checksum feature.
session-protocol	Session protocol to be used for Internet calls between local and remote router via the IP backbone.
req-qos	Configured requested quality of service for calls for this dial peer.
acc-qos	Configures acceptable quality of service for calls for this dial peer.
fax-rate	Configured facsimile transmission speed for with this dial peer.
codec	CODEC type configured for this dial peer.
Expect factor	Configured value at which the system will generate an SMTP message alerting that the voice quality has dropped.
Icpif	Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer.
VAD	Whether or not voice activation detection (VAD) is enabled for this dial peer.
Poor QOV Trap	Whether Poor Quality of Voice trap messages have been enabled or disabled.
Connect Time	Unit of measure indicating the call connection time associated with this dial peer.
Charged Units	Number of call units charged to this dial peer.
Successful Calls	Number of completed calls to this peer since system startup.
Failed Calls	Number of uncompleted (failed) calls to this peer since system startup.
Accepted Calls	Number of calls from this peer accepted since system startup.
Refused Calls	Number of calls from this peer refused since system startup.
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.
Matched	Destination pattern matched for this dial peer.
Target	Matched session target (IP address or host name) for this dial peer.

Field	Description
Dest Digit Pattern	Index number identifying the destination telephone number digit pattern.
Translation	Expanded destination telephone number digit pattern.

Field	Description
(orig)	Indicates the call is originating on the voice port.
eecm	Status of the End-to-End Call Manager.
LFXS	Status of the FXS line.
CPD	Status of the Call Processing Data.
LFXO	Status of the FXO line.
LEM	Status of the E&M line.

Field	Description
DSP	Number of the DSP
Channel	Number of the channel and its status.

slot/port	(Optional) Displays information for only the voice port you specify with the slot/port designation. slot specifies the slot number in the Cisco router where the voice interface card is installed. The only valid entry is 1. port specifies the voice port number. Valid ranges are as follows: Analog voice ports: from 1 to 6. Digital voice port: Digital T1: from 1 to 24. Digital E1: from 1 to 15, and from 17 to 31.
summary	(Optional) Display a summary of all voice ports.

Field	Description
Administrative State	Administrative state of the voice port.
Alias	User-supplied alias for this voice port.
Analog interface A-D gain offset	Offset of the gain for analog-to-digital conversion.
Analog interface D-A gain offset	Offset of the gain for digital-to-analog conversion.
Clear Wait Duration Timing	Time of inactive seizure signal to declare call cleared.
Coder Type	Voice compression mode used.
Companding Type	Companding standard used to convert between analog and digital signals in PCM systems.
Connection Mode	Connection mode of the interface.
Connection Number	Full E.164 telephone number used to establish a connection with the trunk or PLAR mode.
Currently Processing	Type of call currently being processed: none, voice, or fax.
Delay Duration Timing	Maximum delay signal duration for delay dial signaling.
Delay Start Timing	Timing of generation of delayed start signal from detection of incoming seizure.
Description	Description of the voice port.
Dial Type	Out-dialing type of the voice port.
Digit Duration Timing	DTMF Digit duration in milliseconds.
E&M Type	Type of E&M interface.
Echo Cancel Coverage	Echo Cancel Coverage for this port.
Echo Cancellation	Whether or not echo cancellation is enabled for this port.
Hook Flash Duration Timing	Maximum length of hook flash signal.
Hook Status	Hook status of the FXO/FXS interface.
Impedance	Configured terminating impedance for the E&M interface.
In Gain	Amount of gain inserted at the receiver side of the interface.
In Seizure	Incoming seizure state of the E&M interface.
Initial Time Out	Amount of time the system waits for an initial input digit from the caller.
InterDigit Duration Timing	DTMF interdigit duration in milliseconds.
InterDigit Pulse Duration Timing	Pulse dialing interdigit timing in milliseconds.
Interdigit Time Out	Amount of time the system waits for a subsequent input digit from the caller.
Maintenance Mode	Maintenance mode of the voice port.
Maximum Playout Delay	The amount of time before the Cisco MC3810 DSP starts to discard voice packets from the DSP buffer.
Music On Hold Threshold	Configured Music-On-Hold Threshold value for this interface.
Noise Regeneration	Whether or not background noise should be played to fill silent gaps if VAD is activated.
Nominal Playout Delay	The amount of time the Cisco MC3810 DSP waits before starting to play out the voice packets from the DSP buffer.
Non-Linear Processing	Whether or not non-linear processing is enabled for this port.
Number of signaling protocol errors	Number of signaling protocol errors.
Operations State	Operation state of the port.
Operation Type	Operation of the E&M signal: two-wire or four-wire.
Out Attenuation	Amount of attenuation inserted at the transmit side of the interface.
Out Seizure	Outgoing seizure state of the E&M interface.
Port	Port number for this interface associated with the voice interface card.
Pulse Rate Timing	Pulse dialing rate in pulses per second (pps).
Region Tone	Configured regional tone for this interface.
Ring Active Status	Ring active indication.
Ring Cadence	Configured ring cadence for this interface.
Ring Frequency	Configured ring frequency for this interface.
Ring Ground Status	Ring ground indication.
Ringing Time Out	Ringing time out duration.
Signal Type	Type of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial.
Slot	Slot used in the voice interface card for this port.
Sub-unit	Subunit used in the voice interface card for this port.
Tip Ground Status	Tip ground indication.
Type of VoicePort	Type of voice port: FXO, FXS, and E&M.
The Interface Down Failure Cause	Text string describing why the interface is down,
Voice Activity Detection	Whether Voice Activity Detection is enabled or disabled.
Wait Release Time Out	The time that a voice port stays in the call-failure state while the Cisco MC3810 sends a busy tone, reorder tone, or an out-of-service tone to the port.
Wink Duration Timing	Maximum wink duration for wink start signaling.
Wink Wait Duration Timing	Maximum wink wait duration for wink start signaling.

Table of Contents

Voice-Related Commands

Syntax Description

Default

Command Mode

Usage Guidelines