Multiservice Route Processor Commands

Syntax Description

best-effort	Resource Reservation Protocol (RSVP) makes no bandwidth reservation.
controlled-load	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	RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded.

Defaults

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

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Use the acc-qos dial-peer command to generate an SNMP event if the QoS for a 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 QoS can be provided. The IOS software uses RSVP to request QoS 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 IOS software generates a trap message when the bandwidth required to provide the selected QoS is not available.

This command only applies to voice-over-IP (VoIP) peers.

Examples

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

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

Related Commands

Command	Description
req-qos	Specifies the desired quality of service to be used in reaching a specified dial peer in VoIP.

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: Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered. Plus sign (+), which is optionally used as the first digit to indicate an E.164 standard number. Comma (,), which inserts a pause between digits. Period (.), which is used as a wild-card character and matches any entered digit.

Defaults

Enabled with a null string.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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 dial peer associated with the interface is associated with the incoming call.

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

This command applies to both voice-over-IP (VoIP) and POTS dial peers.

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

Examples

The following example configures the E.164 telephone number 14085559626 as the dial peer of an incoming call:

dial-peer voice 10 pots
 answer-address 14085559626

Related Commands

Command	Description
destination-pattern	Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
port	Associates a dial peer with a specific voice port.
prefix	Specifies the prefix of the dialed digits for this dial peer.
timeouts interdigit	Configures the interdigit timeout value for a specified voice port.

Syntax Description

number	Channel group number. When configuring a T1 data line, channel group numbers can be values from 0 to 23.
timeslots range	One or more timeslots or ranges of timeslots belonging to the channel group. The first timeslot is numbered 1. For a T1 controller, the timeslot range is from 1 to 24. For an E1 controller, the timeslot range is from 1 to 31.
speed {64}	(Optional) Line speed (in kilobits per second) of the T1 link.

Defaults

64 kbps

Command Modes

Controller configuration

Usage Guidelines

Use this command in configurations where the MRP must communicate with a T1 fractional data line. The channel group number can be arbitrarily assigned and must be unique for the controller. The timeslot range must match the timeslots assigned to the channel group. The service provider defines the timeslots that comprise a channel group.

Examples

The following example defines three channel groups. Channel group 0 consists of a single timeslot, channel group 8 consists of 7 timeslots and runs at a speed of 64 kbps per timeslot, and channel group 12 consists of two timeslots.

channel-group 0 timeslots 1
channel-group 8 timeslots 5,7,12-15,20 speed 64
channel-group 12 timeslots 2

Related Commands

Command	Description
framing	Specifies the type of framing used by the T1 channels.
linecode	Specifies the type of line coding used by the T1 channels.
ds0-group	Defines the timeslots that belong to each T1voice circuit.

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.

Defaults

g729r8 (G.729 8000 bps)

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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-command values for the voice-over-IP (VoIP) peers of a connection do not match, the call fails.

This command only applies to VoIP peers.

Examples

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

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

Examples

The following example enables background noise:

voice port 0/0
 comfort-noise

Related Commands

Command	Description
vad (dial-peer configuration)	Enables VAD for the calls using a particular dial peer.
vad (voice-port configuration)	Enables VAD for the calls using a particular voice port.

Syntax Description

plar	Private line auto ringdown (PLAR) connection. PLAR connection associates a dial peer directly with an interface; when an interface goes off-hook, the dial peer sets up the second call leg and creates a conference call without the caller having to dial any digits.
trunk	Straight tie-line connection to a private branch exchange (PBX).
string	Destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number.

Defaults

No connection.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.1(3)T	This command was introduced.

Usage Guidelines

Use the connection command to specify a connection mode for a specific interface. Use the connection plar command to specify a PLAR interface. The string you configure for this command is used as the called number for all calls coming in over this voice port. The destination dial peer is determined on the basis of this called number.

Use the connection trunk command to specify a straight tie-line connection to a PBX. This command can be used for E&M-to-E&M trunks, FXO-to-FXS trunks, and FXS-to-FXS trunks. Signaling is transported for E&M-to-E&M trunks and FXO-to-FXS trunks; signaling will not be transported for FXS-to-FXS trunks.

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

Examples

The following example selects plar as the connection mode and a destination telephone number of 14085559262:

voice port 0/0
 connection plar 14085559262

The following example selects trunk as the connection mode and a destination telephone number of 14085559262:

voice port 0/0
 connection trunk 14085559262

Related Commands

Command	Description
session protocol	Establishes a session protocol for calls between the local and remote routers through the packet network in VoIP.

Syntax Description

t1	T1 controller.
slot	Slot number on the MRP where the voice interface card (VIC) is installed. Valid entries are 0 or 1, depending on the slot where it has been installed.
port	Voice port. Valid entries are 0 or 1.

Defaults

No T1 controller is configured.

Command Modes

Global configuration

Command History

Release	Modification
10.0	This command was introduced.
10.3	The e1 keyword was added.

Usage Guidelines

This command is used in configurations where the router or access server is intended to communicate with a T1 fractional data line. Additional parameters for the T1 line must be configured for the controller before the T1 circuits can be configured by means of the interface global configuration command.

Examples

The following example configures slot 1, port 0 as a T1 controller:

controller t1 1/0

Related Commands

Command	Description
channel-group	Defines the timeslots that belong to each T1 circuit.
framing	Specifies the type of framing used by the T1 channels.
linecode	Specifies the type of line coding used by the T1 channels.
show controller t1	Displays information about the T1 links.

Syntax Description

argentina	Analog voice interface-related default tone, ring, and cadence setting for Argentina.
australia	Analog voice interface-related default tone, ring, and cadence setting for Australia.
austria	Analog voice interface-related default tone, ring, and cadence setting for Austria.
belgium	Analog voice interface-related default tone, ring, and cadence setting for Belgium.
brazil	Analog voice interface-related default tone, ring, and cadence setting for Brazil.
canada	Analog voice interface-related default tone, ring, and cadence setting for Canada.
china	Analog voice interface-related default tone, ring, and cadence setting for China.
colombia	Analog voice interface-related default tone, ring, and cadence setting for Colombia.
cyprus	Analog voice interface-related default tone, ring, and cadence setting for Cyprus.
czech republic	Analog voice interface-related default tone, ring, and cadence setting for Czech Republic.
denmark	Analog voice interface-related default tone, ring, and cadence setting for Denmark.
finland	Analog voice interface-related default tone, ring, and cadence setting for Finland.
france	Analog voice interface-related default tone, ring, and cadence setting for France.
germany	Analog voice interface-related default tone, ring, and cadence setting for Germany.
greece	Analog voice interface-related default tone, ring, and cadence setting for Greece.
hongkong	Analog voice interface-related default tone, ring, and cadence setting for Hongkong.
hungary	Analog voice interface-related default tone, ring, and cadence setting for Hungary.
iceland	Analog voice interface-related default tone, ring, and cadence setting for Iceland.
india	Analog voice interface-related default tone, ring, and cadence setting for India.
indonesia	Analog voice interface-related default tone, ring, and cadence setting for Indonesia.
ireland	Analog voice interface-related default tone, ring, and cadence setting for Ireland.
israel	Analog voice interface-related default tone, ring, and cadence setting for Israel.
italy	Analog voice interface-related default tone, ring, and cadence setting for Italy.
japan	Analog voice interface-related default tone, ring, and cadence setting for Japan.
korea republic	Analog voice interface-related default tone, ring, and cadence setting for Korea Republic.
luxembourg	Analog voice interface-related default tone, ring, and cadence setting for Luxembourg.
malaysia	Analog voice interface-related default tone, ring, and cadence setting for Malaysia.
mexico	Analog voice interface-related default tone, ring, and cadence setting for Mexico.
netherlands	Analog voice interface-related default tone, ring, and cadence setting for Netherlands.
new zealand	Analog voice interface-related default tone, ring, and cadence setting for New Zealand.
norway	Analog voice interface-related default tone, ring, and cadence setting for Norway.
peru	Analog voice interface-related default tone, ring, and cadence setting for Peru.
philipines	Analog voice interface-related default tone, ring, and cadence setting for Philipines.
portugal	Analog voice interface-related default tone, ring, and cadence setting for Portugal.
russian federation	Analog voice interface-related default tone, ring, and cadence setting for Russian Federation.
singapore	Analog voice interface-related default tone, ring, and cadence setting for Singapore.
slovakia	Analog voice interface-related default tone, ring, and cadence setting for Slovakia.
slovenia	Analog voice interface-related default tone, ring, and cadence setting for Slovenia.
south africa	Analog voice interface-related default tone, ring, and cadence setting for South Africa.
spain	Analog voice interface-related default tone, ring, and cadence setting for Spain.
sweden	Analog voice interface-related default tone, ring, and cadence setting for Sweden.
switzerland	Analog voice interface-related default tone, ring, and cadence setting for Switzerland.
taiwan	Analog voice interface-related default tone, ring, and cadence setting for Taiwan.
thailand	Analog voice interface-related default tone, ring, and cadence setting for Thailand.
turkey	Analog voice interface-related default tone, ring, and cadence setting for Turkey.
unitedkingdom	Analog voice interface-related default tone, ring, and cadence setting for the United Kingdom.
united states	Analog voice interface-related default tone, ring, and cadence setting for United States.
venezuela	Analog voice interface-related default tone, ring, and cadence setting for Venezuela.

Defaults

united states

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

The following example configures United States as the call progress tone locale:

voice port 0/0
 cptone united states

Syntax Description

string

Character string from 1 to 255 characters.

Defaults

Enabled with a null string.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

The following example identifies this voice port as a connection to the purchasing department:

voice port 0/0
 description purchasing_dept

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 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. 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). 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.

Defaults

Enabled with a null string.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

This command applies to both voice-over-IP (VoIP) and basic telephone service, or plain old telephone service (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 added to the front of the remaining numbers, creating a dial string that the router then dials. If all numbers in the destination pattern are stripped out, the user receives a dial tone.

There are 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 IOS software does not check the validity of the E.164 telephone number; it accepts any series of digits as a valid number.

Examples

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

Command	Description
answer-address	Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call.
prefix	Specifies the prefix of the dialed digits for this dial peer.
timeouts interdigit	Configures the interdigit timeout value for a specified voice port.

Syntax Description

max-size number	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	Number of minutes for retaining entries in the call history table. Valid entries are from 0 to 2147483647 minutes. A value of 0 prevents any history from being retained.

Defaults

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

Command Modes

Global configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

The call history table contains a listing of all calls connected through the router in descending time order since Voice over IP (VoIP) was enabled. Use the dial-control-mib global configuration command to specify attributes for the call history table.

Examples

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

Syntax Description

number	Digits defining a particular dial peer. Valid entries are from 1 to 2147483647.
voip	Voice-over-IP (VoIP) dial peer using voice encapsulation on the basic telephone service, or plain old telephone service (POTS) network.
pots	POTS dial peer using VoIP encapsulation on the IP backbone.

Defaults

No dial peer configuration mode is preconfigured.

Command Modes

Global configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

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

configure terminal
 dial-peer voice 10 pots

Related Commands

Command	Description
voice-port	Opens voice-port configuration mode.

Syntax Description

dtmf	Touch-tone dialer.
pulse	Pulse dialer.

Defaults

dtmf

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

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

voice-port 1/0/0
 dial-type dtmf
 

The following example configures a voice port to support a rotary (pulse tone) dialer:

voice-port 1/1
 dial-type pulse

Syntax Description

number	Channel group number. When configuring a T1 voice line, channel group numbers can be values from 0 to 23.
timeslots range	One or more timeslots or ranges of timeslots belonging to the channel group. The first timeslot is numbered 1. The timeslot range is from 1 to 24.
type	Type of signaling for the voice port. Signaling types include e&m-immediate-start, e&m-delay-dial, e&m-wink-start, fxs-ground-start, fxs-loop-start, fxo-ground-start, and fxo-loop-start.

Defaults

No channel group number is assigned.

Command Modes

Controller configuration

Usage Guidelines

Use this command in configurations where the MRP must communicate with a T1 line. The channel group number may be arbitrarily assigned and must be unique for the controller. The timeslot range must match the timeslots assigned to the channel group. The service provider defines the timeslots that comprise a channel group.

Examples

The following example defines one channel group. Channel group 1 consists of 24 timeslots.

ds0-group 1 timeslots 1-24 type e&m-wink

Related Commands

Command	Description
framing	Specifies the type of framing used by the T1 channels.
linecode	Specifies the type of line coding used by the T1 channels.
channel-group	Defines the timeslots that belong to each T1 data circuit.

Syntax Description

value

Number of milliseconds the echo canceller covers on a given signal. Valid values are 8, 16, 24, and 32 milliseconds.

Defaults

16 milliseconds

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

There is no echo or echo cancellation on the network (non-basic telephone service) side of the connection.

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

Examples

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

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

Related Commands

Command	Description
echo-cancel enable	Enables the cancellation of voice that is sent out the interface and is received on the same interface.

Syntax Description

This command has no arguments or keywords.

Defaults

Enabled for all interface types.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

The echo-cancel enable command enables cancellation of voice signals that is sent from the interface and is transmitted back to the same interface. Sound that is transmitted back in this manner is perceived by the listener as an echo. Disabling echo cancellation might cause the remote side of a connection to generate an echo. Because echo cancellation is an invasive process that can minimally degrade voice quality, this command should be disabled if it is not needed.

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

There is no echo path for a four-wire receive-and-transmit (E&M) interface. The echo canceller should be disabled for that type of interface.

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

Examples

The following example enables the echo cancellation feature and adjusts the size of the echo canceller to 16 milliseconds:

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

Related Commands

Command	Description
echo-cancel coverage	Adjusts the size of the echo canceller.
non-linear	Enables nonlinear processing in the echo canceller.

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.

Defaults

10

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

This command only applies to VoIP peers.

Examples

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

dial-peer voice 10 voip
 expect-factor 0

Syntax Description

2400	Fax transmission speed of 2400 bits per second (bps).
4800	Fax transmission speed of 4800 bps.
7200	Fax transmission speed of 7200 bps.
9600	Fax transmission speed of 9600 bps.
12000	Fax transmission speed of 12000 bps.
14400	Fax transmission speed of 14,400 bps.
disable	Fax relay transmission capability disabled.
voice	Highest possible transmission speed allowed by voice rate.

Defaults

voice

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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 Resource Reservation Protocol (RVSP). Because more network bandwidth will be monopolized by the fax transmission, we do not recommend setting the fax-rate value higher than the codec value. If the fax-rate value is set lower than the codec value, faxes will take longer to transmit but will use less bandwidth.

This command only applies to voice-over-IP (VoIP) peers.

Examples

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

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

Syntax Description

esf	Extended superframe is used as the T1 framing type.
sf	Superframe is used as the T1 framing type.

Defaults

esf

Command Modes

Controller configuration

Command History

Release	Modification
11.3	This command was introduced.

Usage Guidelines

If you do not specify the framing command, the default esf is used.

Examples

The following example sets the framing for the T1 channel to sf:

controller t1 0/0
framing sf

Related Commands

Command	Description
channel-group	Defines the timeslots that belong to each T1 or E1 circuit.
linecode	Specifies the type of line coding used by the T1 channels.
show controller t1	Displays information about the T1 links.

Syntax Description

number

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

Defaults

30 equipment impairment factor units.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

This command only applies to voice-over-IP (VoIP) peers.

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

Examples

The following example disables the icpif command:

dial-peer voice 10 voip
 icpif 0

Syntax Description

600c	600 ohms complex.
600r	600 ohms real.
900c	900 ohms complex.
complex1	Complex 1.
complex2	Complex 2.

Defaults

600r

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Use the impedance command to specify the terminating impedance of a foreign-exchange-office (FXO) voice-port interface. The impedance value selected needs to match the specifications from the specific telephony system to which it is connected. Different countries often have different standards for impedance. Central office (CO) switches in the United States are predominantly 600r. Private branch exchanges (PBXs) in the United States are normally either 600r or 900c.

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

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

This command applies to FXS, FXO, and receive-and-transmit (E&M) voice ports.

Examples

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

voice-port 1/0
 impedance 600r
 

Syntax Description

value

Amount of gain in decibels (dB) to be inserted at the receiver side of the interface. Acceptable value is any integer from -6 to 14.

Defaults

0 dB

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

You cannot increase the gain of a signal going out into the Public Switched Telephone Network (PSTN), but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.

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

Examples

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

port 1/0
 input gain 3
 

Related Commands

Command	Description
output attenuation	Configures a specific output attenuation value for a voice port.

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.

Defaults

No precedence (0).

Command Modes

Dial-peer configuration

Usage Guidelines

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

This command only applies to voice-over-IP (VoIP) peers.

Examples

The following example sets the IP precedence at 5:

dial-peer voice 10 voip
 ip precedence 5

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

This command only applies to voice-over-IP (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 from being forwarded to the DSP.

Examples

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

dial-peer voice 10 voip
 ip udp checksum

Related Commands

Command	Description
loop-detect	Enables loop detection for T1 for Voice over Asynchronous Transfer Mode (ATM), Voice over Frame Relay, and Voice over High-Level Data Link Control (HDLC).

Syntax Description

ami	Specifies that alternate mark inversion (AMI) line coding is used by the T1 channel.
b8zs	Specifies that bipolar 8 zero suppression (B8ZS) line coding is used by the T1 channel.

Defaults

b8zs

Command Modes

Controller configuration

Command History

Release	Modification
11.3	This command was introduced.

Usage Guidelines

Specifies the type of line coding used by the T1 channel.

Examples

The following example sets the line coding for the T1 channel to b8zs:

controller t1 0/0
linecode b8zs

Related Commands

Command	Description
channel-group	Defines the timeslots that belong to each T1 circuit.
framing	Specifies the type of framing used by the T1 channels.
show controller t1	Displays information about the T1 links.

Syntax Description

number

On-hold music threshold in dB. Valid entries are any integer from -70 to -30.

Defaults

-38 dB

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

Examples

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

voice-port 1/0
 music-threshold -35

Syntax Description

This command has no arguments or keywords.

Defaults

Enabled for all voice-port types.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

Examples

The following example enables nonlinear call processing:

voice-port 1/0
 non-linear
 

Related Commands

Command	Description
echo-cancel enable	Enables the cancellation of voice that is sent out the interface and is received on the same interface.

Syntax Description

extension-number	Digits defining an extension number for a particular dial peer.
expanded-number	Digits defining the expanded telephone number or destination pattern for the extension number listed.

Defaults

No number expansion is defined.

Command Modes

Global configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Use the num-exp 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 by 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 wild card—if you want to replace four numbers in an extension with wild cards, enter four periods.

Examples

The following example expands the extension number 54001 to 14085554001:

num-exp 54001 14085554001
 

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

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

Syntax Description

2-wire	Two-wire E&M cabling scheme.
4-wire	Four-wire E&M cabling scheme.

Defaults

2-wire

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

This command does not apply to foreign-exchange-station (FXS) or foreign-exchange-office (FXO) interfaces, because they are, by definition, two-wire interfaces.

Examples

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

voice-port 1/0
 operation 4-wire
 

Syntax Description

value

Amount of attenuation in dB at the transmit side of the interface. Acceptable value is any integer from 0 to 14.

Defaults

The default value for foreign-exchange-office (FXO), foreign-exchange-station (FXS), and receive-and-transmit (E&M) ports is 0 dB.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

You cannot increase the gain of a signal going out into the Public Switched Telephone Network (PSTN), but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.

Examples

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

voice-port 1/0
 output attenuation 3
 

For analog:

For digital:

Syntax Description

slot-number	Slot number in the router where the VIC is installed. Valid entries are 0 or 1, depending on the slot where it has been installed.
port number	Voice port number. Valid entries are 0 or 1.
channel	Channel group number. When configuring a T1 line, channel group numbers can be values from 0 to 23.

Defaults

No port is configured.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

This command only applies to basic telephone service, or plain old telephone service (POTS) peers.

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 voice-over-IP (VoIP) network to match a port with the selected outgoing dial peer.

Examples

The following example associates a dial peer with voice port 1 and accessed through slot 0:

dial-peer voice 10 pots
 port 1/0

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 (,) to indicate a pause in the prefix.

Defaults

Null string.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

This command only applies to basic telephone service, or plain old telephone service (POTS) peers.

Examples

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

dial-peer voice 10 pots
 prefix 9,

Related Commands

Command	Description
answer-address	Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call.
destination-pattern	Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.

Syntax Description

best-effort	Resource Reservation Protocol (RSVP) makes no bandwidth reservation.
controlled-load	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	RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded.

Defaults

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

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

This command only applies to voice-over-IP (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 IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. IOS software uses RSVP to request quality of service guarantees from the network.

Examples

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

Command	Description
acc-qos	Generates an Simple Network Management Protocol (SNMP) event if the quality of service for a dial peer drops below a specified level.

Syntax Description

number

Ring frequency in Hz used in the FXS interface. Valid entries are 25 and 50.

Defaults

25 Hz

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

Examples

The following example configures the ring frequency for 25 Hz:

voice-port 1/0
 ring frequency 25

Related Commands

Command	Description
ring cadence	Specifies the ring cadence for an FXS voice port.
ring number	Specifies the number of rings for a specified foreign-exchange-office (FXO) voice port.

Syntax Description

number

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

Defaults

One ring.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

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

This command does not apply to foreign-exchange-station (FXS) or receive-and-transmit (E&M) interfaces because they do not receive ringing to receive a call.

Examples

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

voice-port 1/0
 ring number 5
 

Related Commands

Command	Description
ring frequency	Specifies the ring frequency for a specified FXS voice port.

Syntax Description

protocol

Cisco Session Protocol.

Defaults

cisco

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

For this release, cisco is the only applicable session protocol. This command only applies to voice-over-IP (VoIP) peers.

Examples

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

dial-peer voice 10 voip
 session protocol cisco

Related Commands

Command	Description
session target	Specifies a network-specific address for a specified dial peer.

Syntax Description

ipv4:destination-address	IP address of the dial peer.
dns:host-name	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 wild cards with this keyword when defining the session target for VoIP peers: $s$.—Source destination pattern will be used as part of the domain name. $d$.—Destination number will be used as part of the domain name. $e$.—Digits in the called number will be reversed, periods will be added in-between each digit of the called number, and this string will be used as part of the domain name. $u$.—Unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name.
loopback:rtp	All voice data will be looped back to the originating source. This only applies to VoIP peers.
loopback:compressed	All voice data will be looped back in compressed mode to the originating source. This only applies to basic telephone service, or plain old telephone service (POTS) peers.
loopback:uncompressed	All voice data will be looped-back in uncompressed mode to the originating source. This only applies to POTS peers.

Defaults

Enabled with no IP address or domain name defined.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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 wild cards. Using the optional wild cards can reduce the number of voice-over-IP (VoIP) dial peer session targets that 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 $u$. wild card. In this example, the destination pattern has been configured to allow for any 4-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. means that the router will use the unmatched portion of the dialed number—in this case, the 4-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 $d$. wild card. In this example, the destination pattern has been configured for 13102221111. The optional wild card $d$. means 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

Related Commands

Command	Description
destination-pattern	Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
session protocol	Establishes a session protocol for calls between the local and remote routers through the packet network in Voice over IP.

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

These two call legs are associated by the connection ID. The connection ID is global across the voice network, so 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.

Examples

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= 

Related Commands

Command	Description
show call history voice	Displays the VoIP call history table.
show dial-peer voice	Displays configuration information for dial peers.
show num-exp	Displays how the number expansions are configured in VoIP.
show voice port	Displays configuration information about a specific voice port.

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 Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Use the show call history voice 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.

Examples

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=
 

Related Commands

Command	Description
show call active voice	Displays the voice-over-IP (VoIP) active call table.
show dial-peer voice	Displays configuration information for dial peers.
show num-exp	Displays how the number expansions are configured in VoIP.
show voice port	Displays configuration information about a specific voice port.

The information displayed is generally useful for diagnostic tasks only and performed by technical support people.

Examples

The following is sample output from the show controllers voice command:

7750# show controllers voice

Last tdm clock command --
   tdm clock 0/1 voice import 0/0 line
Last error:1 -- IPM_TDM_NO_ERROR
 
Current TDM configuration ---
Port Conf Stat    Type    Payload      Timing        EPIC   TSA    PLL  Depend
-------------------------------------------------------------------------------
0/0  Yes   Up     T1/E1    both   export loop         0      0      0   0/1 
0/1  Yes  Down    T1/E1    voice  import loop         0    Undef    0   0/0 
1/0  No  
1/1  No  
 
PLL   Status byCLI   Reference
---------------------------------
 0    Set     Yes      0/0
 1    Unset   No    onboard clock
 
EPIC  Status    VIC    Port0  Port1  PLL
----------------------------------------
 0    Fixed    T1/E1     0      0     0
 1    Unfixed  Undef   Undef  Undef   0
 
TSA   Port    PLL
------------------
 0    0/0      0   
 1    Undef  Undef 
 
System registers --- 
    SSR 0:0x9   , SSR 1:0x6   
    WIC/VIC INTR MASK REG:0x0   
    TDM/SPMM1 INTR MASK REG:0x3F  
    TDM/SPMM2 INTR MASK REG:0x1F  
    SCR 0:0x60  , SCR 1:0xC1  , SCR 2:0x50  
    SCR 3:0xF   , SCR 4:0xC3  , SCR 5:0xF   
    SCR 6:0x0   , SCR 7:0x3   , SCR 8:0xC0  
    SCR 9:0x88  , SCR10:0xFF  , SCR11:0xFF  
    SCR12:0xF0  , SCR13:0x18  , SCR14:0x1E  
    SCR15:0x47  
 
EPIC0 Switch registers:STDA 0xFF STDB 0xFF SARA 0xFF SARB 0xFF SAXA 0xFF SAXB 0xFF STCR 
0x3F MFAIR 0x2D
STAR 0x65 OMDR 0xE2 VNSR 0x0 PMOD 0x4C PBNR 0xFF POFD 0xF0 POFU 0x18
PCSR 0x1 PICM 0x0 CMD1 0xA0 CMD2 0x70 CBNR 0xFF CTAR 0x2 CBSR 0x20 CSCR 0x0
 
EPIC1 Switch registers:STDA 0xFF STDB 0xFF SARA 0xFF SARB 0xFF SAXA 0xFF SAXB 0xFF STCR 
0x3F MFAIR 0x3F
STAR 0x25 OMDR 0xE2 VNSR 0x0 PMOD 0x4C PBNR 0xFF POFD 0xF0 POFU 0x18
PCSR 0x1 PICM 0x0 CMD1 0xA0 CMD2 0x70 CBNR 0xFF CTAR 0x2 CBSR 0x20 CSCR 0x0
 
DSP 0 Host Port Interface:
HPI Control Register 0x303
InterfaceStatus 0x2A MaxMessageSize 0x80
RxRingBufferSize 0x6 TxRingBufferSize 0x9
pInsertRx 0x1 pRemoveRx 0x1 pInsertTx 0x4 pRemoveTx 0x4
 
Rx Message 0:
packet_length 8 channel_id 0 packet_id 1 process id 0xEF01
 
Rx Message 1:
packet_length 8 channel_id 0 packet_id 1 process id 0xEF01
 
Rx Message 2:
packet_length 8 channel_id 0 packet_id 1 process id 0xEF01
 
Rx Message 3:
packet_length 8 channel_id 0 packet_id 1 process id 0xEF01
 
Rx Message 4:
packet_length 8 channel_id 0 packet_id 1 process id 0xEF01
 
Rx Message 5:
packet_length 8 channel_id 0 packet_id 1 process id 0xEF01
 
Tx Message 0:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 1:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 2:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 3:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 4:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 5:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 6:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 7:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Tx Message 8:
packet_length 8 channel_id 0 packet_id 131 process id 0xEF01
 
Bootloader 1.8, Appn 0.0
no application firmware has been loaded
 tx outstanding 0, max tx outstanding 40
 ptr 0x0, length 0x0, max length 0x0
 received 0 packets, 0 bytes, 0 gaint packets
 0 drops, 0 no buffers, 0 input errors 0 input overruns
 184614 bytes output, 724 frames output, 0 output errors, 0 output underrun
 0 unaligned frames

Examples

The following is sample output from the show diag command:

router# show diag

Slot 0:
         C7750 1FE VE T Mainboard Port adapter, 5 ports
         Port adapter is analyzed
         Port adapter insertion time unknown
         EEPROM contents at hardware discovery:
         Hardware Revision        :0.0
         PCB Serial Number        :
         Part Number              :00-0000-00
         Fab Version              :04
         EEPROM format version 4
         EEPROM contents (hex):
           0x00:04 FF 40 01 A8 41 00 00 C1 8B 00 00 00 30 30 30
           0x10:30 00 00 00 00 82 00 00 00 00 02 04 FF FF FF FF
           0x20:FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
           0x30:FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
           0x40:FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
           0x50:FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
           0x60:FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
           0x70:FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
  
         Packet Voice DSP Module:
         Slot 0:
         Hardware Revision        :2.2
         Part Number              :73-4870-01
         Board Revision           :01
         Deviation Number         :0-0
         Fab Version              :01
         PCB Serial Number        :ICP0403000W
         RMA Test History         :00
         RMA Number               :0-0-0-0
         RMA History              :00
         Processor type           :02
         Number of DSP's          :5
         Type of DSP              :TMS320C549
         EEPROM format version 4
         EEPROM contents (hex):
         0x00:  04 FF 40 01 D7 41 02 02 82 49 13 06 01 42 30 31
         0x10:  80 00 00 00 00 02 01 C1 8B 49 43 50 30 34 30 33
         0x20:  30 30 30 57 03 00 81 00 00 00 00 04 00 09 02 FF :
  
         Slot 1:
         Hardware Revision        :2.2
         Part Number              :73-4870-01
         Board Revision           :01
         Deviation Number         :0-0
         Fab Version              :01
         PCB Serial Number        :ICP0403000S
         RMA Test History         :00
         RMA Number               :0-0-0-0
         RMA History              :00
         Processor type           :02
         Number of DSP's          :5
         Type of DSP              :TMS320C549
         EEPROM format version 4
         EEPROM contents (hex):
         0x00:  04 FF 40 01 D7 41 02 02 82 49 13 06 01 42 30 31
         0x10:  80 00 00 00 00 02 01 C1 8B 49 43 50 30 34 30 33
         0x20:  30 30 30 53 03 00 81 00 00 00 00 04 00 09 02 FF
  
         WIC Slot 0:
         T1 (2 port) WAN daughter card WAN daughter card
         Hardware revision 0.3   Board revision V7
         Serial number     4294967295    Part number    255-65535-255
         Test history      0xFF          RMA number     255-255-255
         Connector type    PCI
         EEPROM format version 1
         EEPROM contents (hex):
         0x20:  01 22 00 03 FF FF FF FF FF FF FF FF FF FF FF FF
         0x30:  FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
  
         WIC Slot 1:
         Dual FXS Voice Interface Card WAN daughter card
         Hardware revision 1.1   Board revision F0
         Serial number     0014501512    Part number    800-02493-01
         Test history      0x00          RMA number     00-00-00
         Connector type    WAN Module
         EEPROM format version 1
         EEPROM contents (hex):
         0x20:  01 0E 01 01 00 DD 46 88 50 09 BD 01 00 00 00 00
         0x30:  78 00 00 00 99 05 26 01 FF FF FF FF FF FF FF FF

Syntax Description

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.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Use the show dial-peer voice privileged EXEC command to display the configuration for all voice-over-IP (VoIP) and basic telephone service, or plain old telephone service (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.

Examples

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
 

Related Commands

Command	Description
show call active voice	Displays the VoIP active call table.
show call history voice	Displays the VoIP call history table.
show num-exp	Displays how the number expansions are configured in VoIP.
show voice port	Displays configuration information about a specific voice port.

Syntax Description

dial string

Particular destination pattern (telephone number).

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

The following example displays the dial peer associated with the destination pattern of 123:

7750#show dialplan number 123
Macro Exp.:123
  
VoiceEncapPeer123
         information type = voice,
         tag = 123, destination-pattern = \Q123',
         answer-address = \Q', preference=0,
         numbering Type = \Qunknown'
         group = 123, Admin state is up, Operation state is up,
         incoming called-number = \Q', connections/maximum = 0/unlimited,
         application associated:
         type = pots, prefix = \Q',
         session-target = \Q', voice-port = \Q0/1:0',
         direct-inward-dial = disabled,
         digit_strip = enabled,
  
         register E.164 number with GK = TRUE
         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:123   Digits:3
Target:

Related Commands

Command	Description
show dialplan incall number	Pairs different voice ports and telephone numbers together for troubleshooting VoIP.

Syntax Description

dialed-number

(Optional) Dialed number.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Use the show num-exp 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.

Examples

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...'

Related Commands

Command	Description
show call active voice	Displays the Voice over IP active call table.
show call history voice	Displays the Voice over IP call history table.
show dial-peer voice	Displays configuration information for dial peers.
show voice port	Displays configuration information about a specific voice port.

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Usage Guidelines

Use the show tdm clock command to specify the clock sources on all the slots and ports of a T1 interface.

Examples

The following is sample output from the show tdm clock command:

MRP# show tdm clock

Controller  Payload   State    TDM Clock type
0/0           Voice      up    Import 0/1 Line
0/1           Voice      up    Export Line
1/0            Data      up    Export Line
 
Tdm Clock   Sourced           Feed          Back-Up
0               0/1            0/0              0/0

Related Commands

Command	Description
tdm clock	Specifies the clock source on a T1 interface.

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3 MA	This command was introduced.

Usage Guidelines

This command also applies to Voice over Frame Relay, Voice over Asynchronous Transfer Mode (ATM), and Voice over High-Level Data Link Control (HDLC) on the Cisco MC3810.

Examples

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

7750# show voice dsp

Current total analog signalling channels:0
Current max allowed digital timeslot for voice:48
Current number of DSP group:2
  Group 0:
  Current allocated analog signalling channels:0
  Current free analog signalling channels:0
  Current allocated digital signalling channels:24
  Current free digital signalling channels:0
  port type:T1
  Current Available MIPs:300
  port served:
  SPMM  DSPRM  State   Image     D-sig  D-sig     A-sig  A-sig  Mips  Voice
   Dsp   Dsp                  allocate   free  allocate   free  Free   Chan
   0/0     0      UP    FLEX         8      0         0      0   100      0
   0/1     1      UP    FLEX         8      0         0      0   100      0
   0/2     2      UP    FLEX         8      0         0      0   100      0
  Group 1:
  Current allocated analog signalling channels:0
  Current free analog signalling channels:0
  Current allocated digital signalling channels:24
  Current free digital signalling channels:0
  port type:T1
  Current Available MIPs:300
  port served:
  SPMM  DSPRM  State   Image     D-sig  D-sig     A-sig  A-sig  Mips  Voice
   Dsp   Dsp                  allocate   free  allocate   free  Free   Chan
   1/0     0      UP    FLEX         8      0         0      0   100      0
   1/1     1      UP    FLEX         8      0         0      0   100      0
   1/2     2      UP    FLEX         8      0         0      0   100      0

Related Commands

Command	Description
show dial-peer voice	Displays configuration information for dial peers.
show voice port	Displays configuration information about a specific voice port.

Syntax Description

slot-number	Slot number in the multiservice route processor (MRP) where the voice interface card (VIC) is installed. Valid entries are 0 or 1, depending on the slot where it has been installed.
port-number	Voice port number. Valid entries are 0 or 1.

Command Modes

Privileged EXEC

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

The following is sample output from the show voice port command for a receive-and-transmit (E&M) voice port:

7750# show voice port 1/0

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 a foreign-exchange-station (FXS) voice port:

7750# show voice port 1/0

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 a T1 voice port:

7750#show voice port 0/1:0
  
recEive and transMit - 0/1:0
  Type of VoicePort is E&M
  Operation State is DORMANT
  Administrative State is UP
  No Interface Down Failure
  Description is not set
  Noise Regeneration is enabled
  Non Linear Processing is enabled
  Music On Hold Threshold is Set to -38 dBm
  In Gain is Set to 0 dB
  Out Attenuation is Set to 0 dB
  Echo Cancellation is enabled
  Echo Cancel Coverage is set to 8 ms
  Connection Mode is normal
  Connection Number is not set
  Initial Time Out is set to 10 s
  Interdigit Time Out is set to 10 s
  Call-Disconnect Time Out is set to 60 s
  Ringing Time Out is set to 180 s
  Region Tone is set for US
  
  Analog Info Follows:
  Currently processing none
  Maintenance Mode Set to None (not in mtc mode)
  Number of signaling protocol errors are 0
  Impedance is set to 600r Ohm
  
  Voice card specific Info Follows:
  Signal Type is wink-start
  Operation Type is 2-wire
  E&M Type is 1
  Dial Type is dtmf
  In Seizure is inactive
  Out Seizure is inactive
  Digit Duration Timing is set to 100 ms
  InterDigit Duration Timing is set to 100 ms
  Pulse Rate Timing is set to 10 pulses/second
  InterDigit Pulse Duration Timing is set to 750 ms
  Clear Wait Duration Timing is set to 400 ms
  Wink Wait Duration Timing is set to 200 ms
  Wink Duration Timing is set to 200 ms
  Delay Start Timing is set to 300 ms
  Delay Duration Timing is set to 2000 ms
  Dial Pulse Min. Delay is set to 140 ms
7750#

Related Commands

Command	Description
show call active voice	Displays the voice-over-IP (VoIP) active call table.
show call history voice	Displays the VoIP call history table.
show dial-peer voice	Displays configuration information for dial peers.
show num-exp	Displays how the number expansions are configured in VoIP.

Syntax Description

This command has no arguments or keywords.

Defaults

no shutdown.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

When a dial peer is shut down, you cannot initiate calls to that peer. This command applies to both voice-over-IP (VoIP) and basic telephone service, or plain old telephone service (POTS) peers.

Examples

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

configure terminal
 dial-peer voice 10 pots
 shutdown

Syntax Description

This command has no arguments or keywords.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

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

Examples

The following example takes voice port 1/0 offline:

configure terminal 
 voice-port 1/0
 shutdown

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

For FXO and FXS:

For E&M and T1:

Syntax Description

loop-start	Loop start signaling. Used for foreign-exchange-office (FXO) and foreign-exchange-station (FXS) interfaces. With loop start signaling only one side of a connection can hang up. This is the default setting for FXO and FXS voice ports.
ground-start	Ground start signaling. Used for FXO and FXS interfaces. Ground Start allows both sides of a connection to place a call and to hang up.
wink-start	Calling side seizes the line by going off-hook on its E lead, and then waits for a short off-hook wink indication on its M lead from the called side before sending address information as dual tone multi-frequency (DTMF) digits. Used for receive-and-transmit (E&M) tie trunk interfaces. This is the default setting for E&M voice ports.
immediate	Calling side seizes the line by going off-hook on its E-lead and sends address information as DTMF digits. Used for E&M tie trunk interfaces.
delay-dial	Calling side seizes the line by going off-hook on its E-lead. After a timing interval, the calling side looks at the supervision from the called side. If the supervision is on-hook, the calling side starts sending information as DTMF digits; otherwise, the calling side waits until the called side goes on-hook and then starts sending address information. Used for E&M tie trunk interfaces.

Defaults

loop-start for FXO and FXS interfaces; wink-start for E&M interfaces.

Command Modes

Voice-port configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

Configuring the signal command for an FXS or FXO voice port changes the signal value for both voice ports on a voice interface card (VIC).

Note   If you change the signal type for an FXO voice port, you need to move the appropriate jumper in the VIC.

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

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

Examples

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

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

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled.

Command Modes

Dial-peer configuration

Command History

Release	Modification
11.3(1)T	This command was introduced.

Usage Guidelines

This command only applies to VoIP peers.

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 a Simple Network Management Protocol (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.

Examples

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

Command	Description
snmp-server enable traps	Enables a router to send SNMP traps and informs.
snmp trap link-status	Enables SNMP trap messages to be generated when a specific port is brought up or down.

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 Management Information Base (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, 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 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, 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 both of the following keywords: health—Enables Internet Engineering Task Force (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

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 Modes

Global configuration

Usage Guidelines

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. 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. To enable multiple types of traps, you must issue a separate snmp-server enable traps command for each trap type and option.

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.

All trap types used in this command 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 by 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 does 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

Command	Description
snmp enable peer-trap peer-qov	Generates poor-quality-of-voice notification for applicable calls associated with VoIP dial peers.
snmp trap link-status	Enables SNMP trap messages to be generated when a specific port is brought up or down.

Syntax Description

This command contains no arguments or keywords.

Defaults

Enabled.

Command Modes

Voice-port configuration

Usage Guidelines

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

Examples

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

voice port 1/0
 snmp trap link-status

Related Commands

Command	Description
snmp enable peer-trap peer-qov	Generates notification of poor voice quality for applicable calls associated with voice-over-IP (VoIP) dial peers.
snmp-server enable traps	Enables a router to send SNMP traps.

Syntax Description

slot/port	Slot number and port number on the interface. Valid entries are 0 or 1.
voice	T1 is used for only voice.
data	T1 is used for only data.
both	T1 is used for both voice and data.
export	TDM clock is provided by the line to which this port is connected and might export this clock to other T1 ports.
import	TDM clock is provided by the onboard clock or the line clock of one of the other exporting T1 ports.
line	Other end of the circuit (PSTN) provides the clock.
internal	Onboard clock provides the clock.