Command Reference: A through M

This chapter contains commands to configure and maintain Cisco Unified Survivable Remote Site Telephony (SRST) and Cisco Unified SIP SRST. The commands are presented in alphabetical order. Some commands required for configuring Cisco Unified SRST and Cisco Unified SIP SRST may be found in other Cisco IOS command references. Use the command reference primary index or search online to find these commands.

access-code

To configure trunk access codes for each type of line so that the Cisco IP phones can access the trunk lines only during Cisco Unified Communications Manager fallback when the Cisco Unified SRST feature is enabled, use the access-code command in call-manager-fallback configuration mode. To remove the telephone access code configuration from the Cisco IP phones, use the no form of this command.

FXO and EandM Line Types

access-code {fxo | e&m} dial-string

no access-code {fxo | e&m} dial-string

BRI and PRI Line Types

access-code {bri | pri} dial-string [direct-inward-dial]

no access-code {bri | pri} [dial-string] [direct-inward-dial]

Syntax Description

fxo

Enables a Foreign Exchange Office (FXO) interface.

e&m

Enables an analog ear and mouth (E&M) interface.

dial-string

Sets up dial access codes for each specified line type by creating dial peers.

bri

Enables a BRI interface.

pri

Enables a PRI interface.

direct-inward-dial

(Optional) Enables direct inward dialing on a POTS dial peer.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.1(5)YD

Cisco Unified SRST 1.0

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(2)XT

Cisco Unified SRST 2.0

This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.

12.2(8)T

Cisco Unified SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

Usage Guidelines

The access-code command configures trunk access codes for each type of line—BRI, E&M, FXO, and PRI—so that the Cisco IP phones can access the trunk lines during Cisco Unified Communications Manager fallback when Cisco Unified SRST is enabled. This provides systemwide access.


Note


The access-code command creates temporary dial peers during Cisco Unified Communications Manager fallback. In many cases, you may already have the local PSTN ports configured with appropriate access codes provided by dial peers (for example, dial 9 to select an FXO PSTN line), in which case this command is not needed.

The access-code command creates temporary POTS voice dial peers for all the selected types of voice ports during Cisco Unified Communications Manager fallback. Use this command only if your normal network dial-plan configuration prevents you from configuring permanent POTS voice dial peers to provide trunk access for use in the fallback mode. When the access-code command is used, it is important to ensure that all ports covered by the command have valid trunk connections. Selection between ports for outgoing calls is random.

The dial string is used to set up temporary dial peers for each specified line type. If there are multiple lines of the same type, a dial peer is set up for each line. The dial peers are active only during Cisco Unified Communications Manager fallback when the Cisco Unified SRST feature is enabled. The result of this configuration is that all PSTN interfaces of the same type, for example BRI, are treated as equivalent, and any port may be selected to place the outgoing PSTN call. The direct-inward-dial keyword enables you to set direct-inward-dialing access for PRI and BRI trunk lines.

Examples

The following example sets the access-code command for BRI 8:


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# access-code bri 8 direct-inward-dial
The following example sets the access-code
 command for E&M 8:
Router(config)# call-manager-fallback 
Router(config-cm-fallback)# access-code e&m 8

The following example sets the access-code command for FXO 9:


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# access-code fxo 9

The following example sets the access-code command for PRI 9:


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# access-code pri 9 direct-inward-dial

addon

To define whether a phone type supports add-on modules, use the addon command in ephone-type configuration mode. To reset to the default value, use the no form of this command.

addon

no addon

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled (phone type does not support add-on modules).

Command Modes


Ephone-type configuration (config-ephone-type)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)XZ

Cisco Unified CME 4.3 Cisco Unified SRST 4.3

This command was introduced.

12.4(20)T

Cisco Unified CME 7.0 Cisco Unified SRST 7.0

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

This command specifies that add-on modules, such as the Cisco Unified IP Phone 7915 Expansion Module, are supported by the type of phone being added with the phone-type template.

Examples

The following example shows that expansion modules are supported for the phone being added with an ephone-type template:


Router(config)# ephone-type 7965
Router(config-ephone-type)# addon

address (voice emergency response location)

To define the civic address for an ERL that is used for the ALI database upload, use the address command in voice emergency response location mode. To remove this definition, use the no form of the command. This command is optional.

address string

no address

Syntax Description

string

String (1-247 characters) used to identify an ERL’s civic address.

Command Default

The civic address is not defined.

Command Modes


Voice emergency response location configuration (cfg-emrgncy-resp-location)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)XY

Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)

This command was introduced.

12.4(20)T

Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

This command creates a comma separated text entry of the ERL’s civic address. The address information must be entered to conform with the NENA-2 Data Record specifications or the recommendations by the service provider.

Examples

In this example, a civic address is displayed for ERL 60.


voice emergency response location 60
subnet 1 209.165.200.224 255.255.0.0
elin 1 4085550100
name Cookies and More Incorporated,
address I,408,5550100,,11902,,,Main Street,Emerald City,CA,Idina Menzel,1,,,,,,

after-hour exempt (voice register pool)

To specify that a particular voice register pool does not have any of its outgoing calls blocked, even though global system call blocking is enabled, use the after-hours exempt command in voice register pool configuration mode. To return to the default, use the no form of this command.

after-hour exempt

no after-hour exempt

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled (global call blocking remains active, as configured).

Command Modes


Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 Cisco SRST 3.4

This command was introduced.

Usage Guidelines

This command exempts individual Cisco SIP phones and phone extensions in a voice register pool from call blocking.

Call blocking on Cisco IP phones is defined in the following way. First, define one or more patterns of outgoing digits by using the after-hours block pattern command in either telephony-service configuration mode for Cisco Unified CME or in call-manager-fallback configuration mode for Cisco Unified SIP SRST. Next, define one or more time periods during which calls that match those patterns are to be blocked are by using the after-hours date or after-hours day command or both. By default, all Cisco IP phones in a Cisco Unified CME or Cisco Unified SIP SRST system are restricted during the specified time if at least one pattern and at least one time period are defined.

A phone extension is exempt as long as the after-hour exempt command is configured in voice register dn or in voice register pool configuration mode.


Note


The id (voice register pool) command is required before Cisco Unified CME or Cisco Unified SIP SRST can accept registrations. Configure the id (voice register pool) command before any other voice register pool command.

Examples

The following example shows how to configure a SIP phone, specified by the voice register pool command, so that outgoing calls are not blocked:


Router(config)# voice register pool 23
Router(config-register-pool)# after-hour exempt

The following example shows how to specify that outgoing calls from extension 5001 under voice register pool 2 are not blocked:


Router(config)# voice register pool 2
Router(config-register-pool)# number 5001
Router(config-register-pool)# after-hour exempt

after-hours block pattern (call-manager-fallback)

To define a pattern of outgoing digits for call blocking from IP phones, use the after-hours block pattern command in call-manager-fallback configuration mode. To delete a call-blocking pattern, use the no form of this command.

after-hours block pattern pattern-tag pattern [7-24]

no after-hours block pattern pattern-tag

Syntax Description

pattern- tag

Identifier for a call-blocking pattern. Up to 32 call-blocking patterns can be defined in separate commands.

pattern

Outgoing call digits to be matched for blocking.

7-24

(Optional) If the 7-24 keyword is specified, the pattern is always blocked, 7 days a week, 24 hours a day. If the 7-24 keyword is not specified, the pattern is blocked during the days and dates that are defined with the after-hours day and after-hours date commands.

Command Default

No pattern is defined.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco Unified SRST 3.0

This command was introduced.

12.3(4)T

Cisco Unified SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

15.3(2)T

Cisco Unified CME 9.5

This command was modified to include regular expressions as a value for the pattern argument.

Usage Guidelines

Call blocking on IP phones is defined in the following way. First, one or more patterns of outgoing digits are defined using the after-hours block pattern command. Next, one or more time periods during which calls that match those patterns are to be blocked are defined using the after-hours date or after-hours day command or both. By default, all IP phones in a Cisco Unified SRST system are restricted during the specified time if at least one pattern and at least one time period are defined.

Blocked calls return a fast-busy tone to the user for approximately 10 seconds before the call is terminated and the line is returned to on-hook status.

In Cisco Unified SRST 9.5, support for afterhours pattern blocking is extended to regular expression patterns for dial plans on Cisco Unified SIP and Cisco Unified SCCP IP phones. With this support, users can add a combination of fixed dial plans and regular expression-based dial plans.

When a call is initiated after hours, the dialed number is matched against a combination of dial plans. If a match is found, the call is blocked.

To enable regular expression patterns to be included when configuring afterhours pattern blocking, the after-hours block pattern command is modified to include regular expressions as a value for the pattern argument.


Note


In SRST (call-manager-fallback configuration) mode, there is no phone- or pin-based exempt to after-hours call blocking.

Examples

The following example defines pattern 1, which blocks international calls after hours for a Cisco Unified SRST system that requires dialing 9 for external calls:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# after-hours block pattern 1 9011

The following example shows how to configure several afterhours block patterns of regular expressions:


Router# configure terminal
Enter configuration commands, one per line.  End with CNTL/Z.
Router(config)# call-manager-fallback
 
Router(config-telephony)# after-hours block pattern 1 ?
  WORD  Specific block pattern or a regular expression for after-hour block
        pattern
Router(config-cm-fallback)# after-hours block pattern 1 1234
Router(config-cm-fallback)# after-hours block pattern 2 .T
Router(config-cm-fallback)# after-hours block pattern 3 987654([1-3])+
Router(config-cm-fallback)# after-hours block pattern 4 98765432[1-9]
Router(config-cm-fallback)# after-hours block pattern 5 98765(432|422|456)

after-hours date (call-manager-fallback)

To define a recurring period based on date during which outgoing calls that match defined block patterns are blocked on IP phones, use the after-hours date command in call-manager-fallback configuration mode. To delete a defined time period, use the no form of this command.

after-hours date month date start-time stop-time

no after-hours date month date

Syntax Description

month

Abbreviated month. The following abbreviations for month are valid: jan , feb , mar , apr , may , jun , jul , aug , sep , oct , nov , dec .

date

Date of the month. Range is from 1 to 31.

start-time stop-time

Beginning and ending times for call blocking, in an HH:MM format using a 24-hour clock. The stop time must be greater than the start time. The value 24:00 is not valid. If 00:00 is entered as a stop time, it is changed to 23:59. If 00:00 is entered for both start time and stop time, calls are blocked for the entire 24-hour period on the specified date.

Command Default

No time period based on date is defined for call blocking.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco Unified SRST 3.0

This command was introduced.

12.3(4)T

Cisco Unified SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

Call blocking on IP phones is defined in the following way. First, one or more patterns of outgoing digits are defined using the after-hours block pattern command. Next, one or more time periods during which calls that match those patterns are to be blocked are defined using the after-hours date or after-hours day command or both. By default, all IP phones in a Cisco Unified SRST system are restricted during the specified time if at least one pattern and at least one time period are defined.

Call blocking for the time period that is defined in this command recurs annually on the date specified in the command.


Note


In SRST (call-manager-fallback configuration) mode, there is no phone- or pin-based exempt to after-hours call blocking.

Examples

The following example defines January 1 as an entire day on which calls that match the pattern specified in the after-hours block pattern command are blocked:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# after-hours date jan 1 00:00 00:00

after-hours day (call-manager-fallback)

To define a recurring period based on date during which outgoing calls that match defined block patterns are blocked on IP phones, use the after-hours date command in call-manager-fallback configuration mode. To delete a defined time period, use the no form of this command.

after-hours date month date start-time stop-time

no after-hours date month date

Syntax Description

month

Abbreviated month. The following abbreviations for month are valid: jan , feb , mar , apr , may , jun , jul , aug , sep , oct , nov , dec .

date

Date of the month. Range is from 1 to 31.

start-time stop-time

Beginning and ending times for call blocking, in an HH:MM format using a 24-hour clock. The stop time must be greater than the start time. The value 24:00 is not valid. If 00:00 is entered as a stop time, it is changed to 23:59. If 00:00 is entered for both start time and stop time, calls are blocked for the entire 24-hour period on the specified date.

Command Default

No time period based on date is defined for call blocking.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco Unified SRST 3.0

This command was introduced.

12.3(4)T

Cisco Unified SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

Call blocking on IP phones is defined in the following way. First, one or more patterns of outgoing digits are defined using the after-hours block pattern command. Next, one or more time periods during which calls that match those patterns are to be blocked are defined using the after-hours date or after-hours day command or both. By default, all IP phones in a Cisco Unified SRST system are restricted during the specified time if at least one pattern and at least one time period are defined.

Call blocking for the time period that is defined in this command recurs annually on the date specified in the command.


Note


In SRST (call-manager-fallback configuration) mode, there is no phone- or pin-based exempt to after-hours call blocking.

Examples

The following example defines January 1 as an entire day on which calls that match the pattern specified in the after-hours block pattern command are blocked:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# after-hours date jan 1 00:00 00:00

alias (call-manager-fallback)

To provide a mechanism for rerouting calls to telephone numbers that are unavailable during Cisco Unified Communications Manager fallback, use the alias command in call-manager-fallback configuration mode. To disable rerouting of unmatched call destination calls, use the no form of this command.

alias tag number-pattern to alternate-number [preference preference-value] [cfw number timeout timeout-value] [huntstop]

no alias tag number-pattern to alternate-number [preference preference-value] [cfw number timeout timeout-value] [huntstop]

Syntax Description

tag

Identifier for alias rule range. The range is from 1 to 50.

number-pattern

Pattern to match the incoming telephone number. This pattern may include wildcards. A number-pattern can be used multiple times.

to

Connects the tag number pattern to the alternate number.

alternate-number

Alternate telephone number to route incoming calls to match the number pattern. The alternate number has to be a specific extension that belongs to an IP phone that is actively registered on the Cisco Unified SRST router. The alternate telephone number can be used in multiple alias commands.

preference

(Optional) Assigns a dial-peer preference value to the alias. Use with the max-dn command.

preference-value

(Optional) Preference value of the associated dial peer. The range is from 0 to 10.

cfw number

(Optional) Allows users to set call forward busy and call forward no answer to a set string and override globally configured call forward settings.

timeout timeout-value

(Optional) Sets the ring no-answer timeout duration for call forwarding, in seconds. Range is from 3 to 60000. The cfw keyword assumes that you want to forward calls to the same number for both busy and no-answer call forward cases. The timeout-value applies only to the no-answer call forward case. There is no timeout required for a busy call forward situation.

huntstop

(Optional) Stops call hunting after trying the alternate-number . This keyword only has an effect if the global command no huntstop is set.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(2)XT

Cisco Unified SRST 2.0

This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(8)T

Cisco Unified SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco Unified SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco Unified SRST 2.01

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers. The preference keyword was added.

12.3(11)T

Cisco Unified SRST 3.2

The cfw keyword was added. The maximum number of alias commands used for creating calls to telephone numbers that are unavailable during Cisco Unified Communications Manager fallback was increased to 50. The alternate-number argument can be used in multiple alias commands.

Usage Guidelines

The alias command provides a mechanism for rerouting calls to telephone numbers that are unavailable during fallback. Up to 50 sets of rerouting alias rules can be created for calls to telephone numbers that are unavailable during Cisco Unified Communications Manager fallback. Sets of alias rules are created using the alias command. An alias is activated when a telephone registers that has a phone number matching a configured alternate-number alias. Under that condition, an incoming call is rerouted to the alternate number. The alternate-number argument can be used in multiple alias commands, allowing you to reroute multiple different numbers to the same target number.

The configured alternate-number must be a specific E.164 phone number or extension that belongs to an IP phone registered on the Cisco Unified SRST router. When an IP phone registers with a number that matches an alternate-number , an additional POTS dial peer is created. The destination pattern is set to the initial configured number-pattern , and the POTS dial peer voice port is set to match the voice port associated with the alternate-number .

If other IP phones register with specific phone numbers within the range of the initial number-pattern , the call is routed back to the IP phone rather than to the alternate-number (according to normal dial-peer longest-match, preference, and huntstop rules).

The cfw keyword allows you to configure a call forward destination for calls that are busy or not answered. Call forward no answer is defined as when the phone rings for a few seconds and the call is not answered, it is forwarded to the configured destination. Call forward busy and call forward no answer can be configured to a set string and override globally configured call forward settings.


Note


Globally configured settings are selected under call-manager-fallback and apply to all phones that register for SRST service.

You can also create a specific call forwarding path for a particular number. The benefit of using the cfw keyword is that during SRST, you can reroute calls from otherwise unreachable numbers onto phones that are available. Basic hunt groups can be established with call-forwarding rules so that if the first SRST phone is busy, you can forward the call to a second SRST phone.

The cfw keyword also allows you to alias a phone number to itself, permitting setting of per-phone number forwarding. An example of aliasing a number to itself follows. If a phone registers with extension 1001, a dial peer that routes calls to the phone is automatically created for 1001. If the call-manager-fallback dial-peer preference (set with the max-dn command) for this initial dial peer is set to 2, the dial peer uses 2 as its preference setting.

Then, use the alias command to alias the phone number to itself:


alias 1 1001 to 1001 preference 1 cfw 2001 timeout 20 

In this example, you have created a second dial peer for 1001 to route calls to 1001, but that has preference 1 and call forwarding to 2001. Because the preference on the dial peer created by the alias command is now a lower numeric value than the preference that the dial peer first created, all calls come initially to the dial peer created by the alias command. In that way the calls are subject to the forwarding as set by the alias command, instead of any call forwarding that may have been set globally.

The alias huntstop keyword is relevant only if you have also set the global no huntstop command under call-manager-fallback. Also, you may need to set the global no huntstop if you have multiple alias commands with the same number-pattern , and you want to enable hunting on busy between the aliases. That is, one alias for number-pattern is tried, and then if that phone is busy, the second alias for number-pattern is tried.

The alias huntstop keyword allows you to turn huntstop behavior back on for an individual alias, if huntstop is turned off globally by the no huntstop command. Setting the huntstop keyword on an individual alias stops hunting at the alias, making the alias the final member of the hunt sequence.


Note


The alias command supports all port types and obsoletes the default-destination command. The alias command is recommended over the default-destination command.

Examples

In the following example, alias 1 is configured to route calls to extensions 6000 through 6099 to extension 5001 using a dial peer with a preference value of 2. Extensions 6000 through 6099 are a subset of IP phones without fallback service. During fallback, calls to these extensions are routed to 5001.


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# alias 1 60.. to 5001 preference 2

In the following example, alias 1 is set up to route calls coming into extensions 5000 through 5999 to extension 6000. The routing occurs when extensions 5000 through 5999 are unavailable during Cisco Unified Communications Manager fallback.


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# alias 1 5... to 6000 

The following example sets the preference keyword in the alias command to a lower preference value than the preference number created by the max-dn command. Setting a lower value allows the cfw keyword to take effect. The incoming call to extension 1000 hunts to alias because it has a lower preference, and no-answer/busy calls to 1000 are forwarded to 2000. All incoming calls to other extensions in SRST mode are forwarded to 3000 after 10 seconds.


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# alias 1 1000 to 1000 preference 1 cfw 2000 timeout 10
Router(config-cm-fallback)# max-dn 10 preference 2
Router(config-cm-fallback)# call-forward busy 3000
Router(config-cm-fallback)# call-forward noan 3000 timeout 10

alias (voice register pool)

To allow Cisco Session Initiation Protocol (SIP) IP phones to handle inbound PSTN calls to telephone numbers that are unavailable when the main proxy is not available, use the alias command in voice register pool configuration mode. To disable rerouting of unmatched call destination calls, use the no form of this command.

alias tag pattern to target [preference value]

no alias tag

Syntax Description

tag

Number from 1 to 5 and the distinguishing factor when there are multiple alias commands.

pattern

Prefix number that represents a pattern against which to match the incoming telephone number. It may include wildcards.

to

Connects the number pattern to the target (alternate number).

target

Target number. An alternate telephone number to route incoming calls that match the number pattern. The target must be a full E.164 number.

preference value

(Optional) Assigns a dial-peer preference value to the alias. The value argument is the value of the associated dial peer. The range is from 1 to 10. There is no default.

Command Default

None

Command Modes


Voice register pool configuration

Command History

Release

Product Version

Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced to Cisco Unified.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

The alias command services calls placed to telephone numbers that are unavailable because the main proxy is not available. The alias command is activated when a Cisco SIP IP phone that has an extension number matching the target number registers.

When a phone with the target number registers, calls that match the number pattern are rerouted to the target number. The target number must be a local phone number to enable rerouting of a range of number patterns. When a Cisco SIP IP phone registers with a target number, an additional VoIP dial peer is created using the target number IP address as a session target and destination pattern as configured with the alias pattern command. For the alias command to work, the VoIP dial peer must be set with a translation rule to translate the called number to the target number. Translation rules can be configured under voice register pool configuration mode.

If other Cisco SIP IP phones register that have specific phone numbers that fall within the alias range or if another static dial peer exists for this pattern, the call is routed using the appropriate dial peer in preference to being rerouted to this alternate alias number (according to normal dial-peer longest-match, preference, and huntstop rules).


Note


The id (voice register pool) command must be configured before any other voice register pool commands, including the alias command. The id command identifies a locally available individual Cisco SIP IP phone or sets of Cisco SIP IP phones. Before the alias command is configured, translation rules must be set using the translate-outgoing (voice register pool) command. Translation rules are a general-purpose number-manipulation mechanism that perform operations such as automatically adding telephone area and prefix codes to dialed numbers.

Examples

The following example configures calls to numbers in the 5000 to 5099 range that are not otherwise explicitly resolved to a specific extension number to be routed to the phone with extension 5001. Phone calls intended for phones that are not given fallback service can then be redirected to the specified extension number.


Router(config)# voice register pool
Router(config-register-pool)# alias 1 50.. to 5001

allow-hash-in-dn(voice register global)

To allow the insertion of '#' at any place in voice register dn, use the allow-hash-in-dn command in voice register global mode. To disable this, use the no form of this command.

allow-hash-in-dn

no allow-hash-in-dn

Syntax Description

allow-hash-in-dn

Allow the insertion of hash at all places in voice register dn.

Command Default

This command is disabled by default.

Command Modes


Voice register global configuration

Command History

Cisco IOS Release

Cisco Product

Modification

15.6(2)T

Cisco SIP SRST 11.0

This command was introduced.

Usage Guidelines

Before this command was introduced, the characters supported in voice register dn were 0-9, +, and *. The new command is enabled whenever the user requires the insertion of # in voice register dn. The command is disabled by default. You can configure this command only in Cisco Unified SRST and Cisco Unified E-SRST modes. The character # can be inserted at any place in voice register dn. When this command is enabled, users are required to change the default termination character(#) to another valid character using ?dial-peer terminator? command under configuration mode.

Examples

The following example shows how to enable the command in mode E-SRST, SRST and how to change the default terminator:


Router(config)#voice register global
Router(config-register-global)#mode esrst
Router(config-register-global)#allow-hash-in-dn

Router(config)#voice register global
Router(config-register-global)#no mode [Default SRST mode]
Router(config-register-global)#allow-hash-in-dn



Router(config)#dial-peer terminator ?
WORD Terminator character: '0'-'9', 'A'-'F', '*', or '#'

Router(config)#dial-peer terminator *

application (call-manager-fallback)

To select the session-level application for all Cisco IP phone lines served by the Cisco Unified SRST router, use the application command in call-manager-fallback configuration mode. To disable this application, use the no form of this command.

application application-name

no application application-name

Syntax Description

application-name

Interactive voice response (IVR) application name.

Command Default

DEFAULT system session application

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco Unified SRST 3.0

This command was introduced.

12.3(4)T

Cisco Unified SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

Use this command to assign a Tool Command Language (Tcl) IVR application to all IP phones served by the Cisco Unified SRST router.

Examples

The following example selects a Tcl IVR application named “app-xfer” for all IP phones served by the Cisco Unified SRST router:


Router(config)# call-manager-fallback
Router(config-cm-fallback) application app-xfer

application (voice register global)

To select the session-level application for all dial peers associated with Session Initiation Protocol (SIP) phones, use the application command in voice register global configuration mode. To disable use of the application, use the no form of this command.

application application-name

no application

Syntax Description

application-name

Interactive voice response (IVR) application name.

Command Default

Default application on router

Command Modes


Voice register global configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was introduced.

Usage Guidelines

During Cisco Unified Communications Manager Express (Cisco Unified CME) or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) registration, a dial peer is created for each SIP phone and that dial peer includes the default session application. The application command allows you to change the default application for all dial peers associated with the Cisco SIP IP phones, if desired. The applied application (or TCL IVR script) must support call redirection. Use the show call application voice summary command to display a list of applications.

The application command in voice register pool configuration mode takes precedence over this command in voice register global configuration mode.


Note


Configure the id (voice register pool) command before any other voice register pool commands, including the application command. The id command identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.

Examples

The following example shows how to set the Tcl IVR application globally for all SIP phones:


Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# application sipapp2

application (voice register pool)

To select the session-level application for the dial peer associated with an individual Session Initiation Protocol (SIP) phone in a Cisco Unified Communications Manager Express (Cisco Unified CME) environment or for a group of phones in a Cisco Unified SIP Survivable Remote Site Telephony (SRST) environment, use the application command in voice register pool configuration mode. To disable use of the application, use the no form of this command.

application application-name

no application

Syntax Description

application-name

Name of the selected interactive voice response (IVR) application name.

Command Default

Default application on router

Command Modes


Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was added to Cisco CME.

Usage Guidelines

During Cisco Unified CME or Cisco Unified SIP SRST registration, a dial peer is created for each SIP phone and that dial peer includes the default session application. The application command allows you to change the default application for all dial peers associated with the Cisco SIP IP phones, if desired. The applied application (or TCL IVR script) must support call redirection. Use the show call application voice summary command to display a list of applications.

The application command in voice register pool configuration mode takes precedence over this command in voice register global configuration mode.


Note


Configure the id (voice register pool) command before any other voice register pool commands, including the application command. The id command identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.

Examples

The following example shows how to set the IVR application for the SIP phone specified by the voice register pool command:


Router(config)# voice register pool 1
Router(config-register-pool) application sipapp2

The following partial sample output from the show running-config command shows that voice register pool 1 has been set up to use the SIP.app application:


voice register pool 1 
 id network 172.16.0.0 mask 255.255.0.0
 application SIP.app
 voice-class codec 1

attempted-registrations size

To set the size of the table that shows a number of attempted-registrations, use the attempted- registrations command in voice register global mode. To set the size of attempted-registrations table to its default value, use the no form of this command.

attempted-registrations size size

no attempted-registrations size

Syntax Description

size

Number of entries in attempted registrations table. Size range from 0 to 50.

Command Default

The default size for attempted registration table is 10.

Command Modes


voice register global

Command History

Cisco IOS Release

Cisco Product

Modification

15.1(2)T

Cisco Unified CME 8.1 Cisco Unified SRST 8.1

This command was introduced.

Usage Guidelines

Use this command to define the size of the table that stores information related to SIP phones that attempt to register with Cisco Unified CME or Cisco Unified SRST and fail. The default size of an attempted registration table is 10. The minimum size of attempted registration table is 0. Use the attempted-registration size 0 when you do not wish to store any information about phones attempting to register with the Cisco Unified CME or Cisco Unified SRST and fail. The maximum size of attempted registration table is 50.

When the current number of entries in the table is more than the new size that is being configured, system prompts the user for the following confirmation, “This will remove x old entries from the table. Proceed? Yes/No ?”. The default user confirmation is “No”. Where “x” represents the number of entries that will be deleted. The old entries are classified on basis of the time-stamp of the latest register attempt made by the phone.

During rollback, the user confirmation is not sought and the target configuration is applied. If the current number of entries in the table is more than the default value of the table size, then entries in excess of the default table size are cleared before reverting to the target table size.

For example, if the configured table size is 40 and there are currently 35 entries in the table, any change in the size of the attempted registration table during rollback removes 25 oldest entries leaving only the default (10) entries before making the table size equal to the size in target configuration.

Examples

The following example shows attempted-registrations size:


Router# conf t
Router(config)#voice register global
Router(config-register-global)#attempted-registrations size 15
!

audible-tone

To configure audible tones to indicate successful join and unjoin and login and logout from any hunt group, use the audible-tone command in ephone or ephone-template configuration mode. To revert to the default behavior of not playing any audible tone, use the no form of this command.

audible-tone

no audible-tone

Syntax Description

This command has no arguments or keywords.

Command Default

By default, this feature is disabled.

Command Modes

Ephone configuration (config-ephone)

Ephone-template configuration (config-ephone-template)

Command History

Cisco IOS Release Cisco Product Modification
15.4(3)M Cisco Unified Enhanced SRST 10.5

This command was introduced.

Usage Guidelines

Use the audible-tone command to set an audible tone to confirm successful join and unjoin and log in and log out from specific hunt groups.

Examples

The following example shows how to configure audible tone in ephone configuration mode:

Router(config)# ephone 1
Router(config-ephone)# audible-tone

The following example shows how to configure audible tone in ephone-template configuration mode:

Router(config)# ephone-template 1
Router(config-ephone-template)# audible-tone

authenticate (voice register global)

To define the authenticate mode for SIP phones in a Cisco Unified CME or Cisco Unified SRST system, use the authenticate command in voice register global configuration mode. To return to the default, use the no form of this command.

Cisco IOS Release 12.4(11)XJ and later releases

authenticate {credential tag location | ood-refer | presence | realm string | register}

no authenticate {credential tag location | ood-refer | presence | realm string | register}

Cisco IOS Release 12.4(4)T

authenticate [all] [realm string]

no authenticate [all] [realm string]

Syntax Description

credential tag

Number that identifies the credential file to use for out-of-dialog REFER (OOD-R) or presence authentication. Range: 1 to 5.

location

Name and location of the credential file in URL format. Valid storage locations are TFTP, HTTP, and flash memory.

ood-refer

Incoming OOD-R requests are authenticated using RFC 2617-based digest authentication.

presence

Incoming presence subscription requests from an external presence server are authenticated.

realm string

Realm parameter for challenge and response as specified in RFC 2617 is authenticated.

register

All incoming registration requests are challenged and authenticated. Valid for Cisco Unified CME only.

Command Default

Authenticate mode is disabled.

Command Modes


Voice register global configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4

This command was introduced.

12.4(11)XJ

Cisco Unified CME 4.1 Cisco Unified SRST 4.1

The credential , ood-refer , presence , and register keywords were added. The register keyword replaced the all keyword.

12.4(15)T

Cisco Unified CME 4.1 Cisco Unified SRST 4.1

The modifications to this command were integrated into Cisco IOS Release 12.4(15)T.

Usage Guidelines

The credential keyword allows OOD-R and presence service to use credential files for authentication. Up to five text files containing username and password pairs can be defined and loaded into the system. The contents of these five files are mutually exclusive; the username and password pairs must be unique across all the files. For Cisco Unified CME, the username and password pairs cannot be the same ones defined for SCCP or SIP phones with the username command.

The ood-refer keyword specifies that any OOD-R request that passes authentication is authorized to setup calls between referee and refer-target if OOD-R is enabled with the refer-ood enable command.

The presence keyword enables digest authentication for external watchers. Credentials are verified against a credential file stored in flash. This applies to both OOD-R and presence. The default is to authenticate all SUBSCRIBE requests from external watchers. An external watcher that passes authentication is authorized to subscribe to presence service for all lines allowed to be watched.

The register keyword enables authentication for registration requests in which the MAC address of the SIP phone cannot be identified by using other methods. All incoming register requests are challenged and authenticated. The realm keyword with the string argument specifies the character string to be included in the challenge.

Examples

The following example shows that all registration requests from SIP phones in a Cisco Unified CME system must be authenticated:


Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# authenticate register

b2bua

To configure a dial peer associated with an individual SIP phone in a Cisco Unified Communications Manager Express (Cisco Unified CME) environment or a group of phones in a Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) environment to point to Cisco Unity Express, use the b2bua command in dial-peer configuration mode. To disable B2BUA call flow on the dial peer, use the no form of this command.

b2bua

no b2bua

Syntax Description

This command has no arguments or keywords.

Command Default

Disabled

Command Modes


Dial-peer configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 and Cisco SIP SRST 3.4

This command was introduced.

Usage Guidelines

Use the b2bua command to set the Cisco Unified CME source address as the 302 redirect contact address for all calls forwarded to Cisco Unity Express.


Note


Use the b2bua command to configure Cisco SIP SRST 3.4 only after using the allow-connections command to enable B2BUA call flow on the SRST gateway.

Examples

The following example shows b2bua included in the configuration for voice dial peer 1:


dial-peer voice 1 voip
 destination-pattern 4...
 session target ipv4:10.5.49.80
 session protocol sipv2
 dtmf-relay sip-notify
 b2bua

call-forward b2bua all

To enable call forwarding for a Session Initiation Protocol (SIP) back-to-back user agent (B2BUA) so that all incoming calls are forwarded to another extension, use the call forward b2bua all command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.

call-forward b2bua all directory-number

no call-forward b2bua all

Syntax Description

directory number

Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.

Command Default

Disabled (no incoming call forwarding to another extension).

Command Modes


Voice register dn configuration
Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was introduced.

Cisco IOS XE Amsterdam 17.2.1r

Cisco Unified SIP SRST 12.8

Introduced support for YANG models.

Usage Guidelines

You can apply call forward to an individual SIP extension (voice register dn) or to the SIP phone on which the extension appears (voice register pool). Use this command in voice register pool configuration mode to enable call forwarding for all extensions on a SIP phone. Use this command in voice register dn configuration mode to enable call forwarding for an individual SIP extension.

If information is configured in both voice register dn and voice register pool mode, the information under voice register dn mode takes precedence.

We recommend that you do not use this command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups unless you want to forward calls to all phones in the hunt group.

The call-forward b2bua all command takes precedence over the call-forward b2bua busy and call-forward b2bua noan commands.


Note


This command in voice register dn configuration mode is not commonly used for Cisco Unified SIP SRST.

Examples

The following example shows how to forward all incoming calls to extension 5001 on directory number 4, to extension 5005.


Router(config)# voice register dn 4
Router(config-register-dn)# number 5001
Router(config-register-dn)# call-forward b2bua all 5005

The following example shows how to forward all incoming calls for extension 5001 on pool number 4, to extension 5005.


Router(config)# voice register pool 4
Router(config-register-pool)# number 5001
Router(config-register-pool)# call-forward b2bua all 5005

call-forward b2bua busy

To enable call forwarding for a Session Initiation Protocol (SIP) back-to-back user agent (B2BUA) so that incoming calls to a busy extension are forwarded to another extension, use the call forward b2bua busy command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.

call-forward b2bua busy directory-number

no call-forward b2bua busy

Syntax Description

directory number

Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.

Command Default

Disabled (no incoming calls to a busy extension are forwarded).

Command Modes


Voice register dn configuration
Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was introduced.

Usage Guidelines

The call-forward b2bua busy response is triggered when a call is sent to a SIP phone using a VoIP dial peer and a busy response is received back from the phone. This command functions only with phones that are registered to a Cisco Unified SIP SRST or Cisco Unified CME router.

You can apply call forward to an individual SIP extension (voice register dn) or to the SIP phone on which the extension appears (voice register pool). Use this command in voice register pool configuration mode to enable call forwarding for all extensions on a SIP phone. Use this command in voice register dn configuration mode to enable call forwarding for a specific extension. If information is configured in both voice register dn and voice register pool mode, the information under voice register dn takes precedence.

We recommend that you do not use this command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.

The call-forward b2bua all command takes precedence over the call-forward b2bua busy and call-forward b2bua noan commands.


Note


This command in voice register dn configuration mode is not commonly used for Cisco Unified SIP SRST.

Cisco Unified CME

Call forward busy can also get invoked if a number is unreachable but the call forward b2bua unreachable command is not configured.

Examples

The following example shows how to forward calls from extension 5001 in pool 4 to extension 5005 when extension 5001 is busy.


Router(config)# voice register pool 4
Router(config-register-pool)# number 5001
Router(config-register-pool)# call-forward b2bua busy 5005

call-forward b2bua mailbox

To control the specific voice-mail box selected in a voice-mail system at the end of a call forwarding exchange, use the call forward b2bua mailbox command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.

call-forward b2bua mailbox directory-number

no call-forward b2bua mailbox

Syntax Description

directory number

Telephone number to which calls are forwarded when the forwarded destination is busy or does not answer. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.

Command Default

Disabled (no voice-mail box is selected for call forwarding).

Command Modes


Voice register dn configuration
Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was introduced.

Usage Guidelines

Use this command to denote the voice-mail box to use at the end of a chain of call forwards to busy or no answer destinations. It can be used to forward calls to a voice-mail box that has a different number than the forwarding extension. A sample of this would be in the case of a shared voice-mail box, for instance one between a manager and her assistant. This command functions only with phones that are registered to a Cisco Unified CME or Cisco Unified SIP SRST router.

If information is configured in both voice register dn and voice register pool mode, the information under voice register dn takes precedence.

It is recommended that you do not use the call-forward b2bua mailbox command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.

This command is used in conjunction with the call-forward b2bua all , call-forward b2bua busy , and call-forward b2bua noan commands.


Note


This command in voice register dn configuration mode is not commonly used for Cisco Unified SIP SRST.

Examples

The following example shows how to forward calls to extension 5005 if an incoming call is forwarded to extension 5001 on pool number 4 and extension 5001 is busy or does not answer.


Router(config)# voice register pool 4
Router(config-register-pool)# number 5001
Router(config-register-pool)# call-forward b2bua mailbox 5005

call-forward b2bua noan

To enable call forwarding for a Session Initiation Protocol (SIP) back-to-back user agent (B2BUA) so that incoming calls to an extension that does not answer after a configured amount of time are forwarded to another extension, use the call forward b2bua noan command in voice register dn or voice register pool configuration mode. To disable call forwarding, use the no form of this command.

call-forward b2bua noan directory-number timeout seconds

no call-forward b2bua noan

Syntax Description

directory number

Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.

timeout seconds

Number of seconds that a call can ring with no answer before the call is forwarded to another extension. Range is 3 to 60000. Default is 20.

Command Default

Disabled (no incoming calls to an extension that does not answer are forwarded).

Command Modes


Voice register dn configuration
Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was introduced.

Usage Guidelines

This command functions only with phones that are registered to a Cisco Unified SIP SRST or Cisco Unified CME router. You can apply call forward to an individual SIP extension (voice register dn) or to the SIP phone on which the extension appears (voice register pool). Use this command in voice register pool configuration mode to enable call forwarding for all extensions on a SIP phone. Use this command in voice register dn configuration mode to enable call forwarding for a specific extension.

If information is configured in both voice register dn and voice register pool mode, the information under voice register dn takes precedence.

We recommend that you do not use this command with hunt groups. If the command is used, consider removing the phone from any assigned hunt groups, unless you want to forward calls to all phones in the hunt group.

The call-forward b2bua all command takes precedence over the call-forward b2bua busy and call-forward b2bua noan commands.


Note


This command in voice register dn configuration mode is not commonly used for Cisco Unified SIP SRST.

Examples

The following example shows how to forward calls to extension 5005 when extension 5001 on pool number 4 is unanswered. The timeout before the call is forwarded to extension 5005 is 10 seconds.


Router(config)# voice register pool 4
Router(config-register-pool)# number 5001
Router(config-register-pool)# call-forward b2bua noan 5005 timeout 10

call-forward busy (call-manager-fallback)

To configure call forwarding to another number when a Cisco IP phone is busy, use the call-forward busy command in call-manager-fallback configuration mode. To disable call forwarding, use the no form of this command.

call-forward busy directory-number

no call-forward busy [directory-number]

Syntax Description

directory-number

Directory number representing a fully qualified E.164 number. This number can contain “.” wildcard characters that correspond to the right-justified digits in the directory number extension.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(2)XT

Cisco Unified SRST 2.0

This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(8)T

Cisco Unified SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco Unified SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco Unified SRST 2.1

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.

Usage Guidelines

The call-forward busy command configures call forwarding to another number when a Cisco IP phone is busy. The call forwarding mechanism is applied globally to all phones that register during fallback.

Examples

The following example forwards calls to extension number 5005 when an incoming call reaches a busy IP phone extension number:


Router(config)# call-manager-fallback
 
Router(config-cm-fallback)# call-forward busy 5005

The following example transforms an extension number for call forwarding when the extension number is busy. The call-forward busy command has an argument of 50.., which prepends the digits 50 to the last two digits of the called extension. The resulting extension is the number to which incoming calls are forwarded when the original extension number is busy. For instance, an incoming call to the busy extension 6002 will be forwarded to extension 5002, and an incoming call to the busy extension 3442 will be forwarded to extension 5042.


Router(config)# call-manager-fallback
 
Router(config-cm-fallback)# call-forward busy 50..

Note


You can forward an incoming VoIP call only to destination numbers local to the router. VoIP calls cannot be forwarded to an alternate (on-net) VoIP destination.

call-forward noan (call-manager-fallback)

To configure call forwarding to another number when no answer is received from a Cisco IP phone, use the call-forward noan command in call-manager-fallback configuration mode. To disable call forwarding, use the no form of this command.

call-forward noan directory-number timeout seconds

no call-forward noan [directory-number]

Syntax Description

directory-number

Directory number representing a fully qualified E.164 number or a local extension number. This number can contain “.” wildcard characters that correspond to the right-justified digits in the directory number extension.

timeout

Sets the ringing no answer timeout duration, which is the waiting time (in seconds) before the call is forwarded to another phone.

seconds

Time (in seconds) before call forwarding starts. The range is from 3 to 60000.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(2)XT

Cisco SRST 2.0

This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(8)T

Cisco SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco SRST 2.1

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.

Usage Guidelines

The call-forward noan command configures call forwarding to another number when no answer is received from a Cisco IP phone. The call-forwarding mechanism is applied globally to all phones that register during fallback. The timeout keyword sets the waiting time before the call is forwarded to another phone. The time is set in seconds. The range is from 3 to 60,000 seconds.

Examples

The following example shows how to set call forwarding of incoming calls to directory number 5005 when line 1, directory number 5001, does not answer. The timeout period before the call is forwarded to directory number 5005 is set for 10 seconds.


Router(config)# call-manager-fallback
Router(config-cm-fallback)# call-forward noan 5005 timeout 10

The following example shows how to set call forwarding of incoming calls to an available extension in the 50xx bank of extensions when line 1, directory number 5001, does not answer. The timeout period before the call is forwarded to directory number 5005 is set for 10 seconds.


Router(config)# call-manager-fallback
Router(config-cm-fallback)# call-forward noan 50.. timeout 10

Note


An incoming VoIP call can be forwarded only to destination numbers local to the router. VoIP calls cannot be forwarded to an alternate (on-net) VoIP destination.

call-forward pattern (call-manager-fallback)

To specify a pattern for calling-party numbers that are able to support the ITU-T H.450.3 standard for call forwarding, use the call-forward pattern command in call-manager-fallback configuration mode. To remove the pattern, use the no form of this command.

call-forward pattern pattern

no call-forward pattern pattern

Syntax Description

pattern

String that consists of one or more digits and wildcard markers or dots (.) to define a specific pattern. Calling parties that match a defined pattern use the H.450.3 standard if they are forwarded. A pattern of .T specifies the H.450.3 forwarding standard for all incoming calls.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco SRST 3.0

This command was introduced.

12.3(4)T

Cisco SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

Usage Guidelines

The pattern match in this command is against the phone number of the calling party. When a directory number has forwarded its calls and an incoming call is received for that number, the SRST router sends an H.450.3 response back to the original calling party to request that the call be placed again using the forward-to destination.

Calling numbers that do not match the patterns defined with this command are forwarded using Cisco-proprietary call forwarding for backward compatibility.

Examples

The following example specifies that all 4-digit directory numbers beginning with 4 should use the H.450.3 standard whenever they are forwarded:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# call-forward pattern 4...

The following example forwards all calls using the H.450.3 standard:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# call-forward pattern .T

call-manager-fallback

To enable Cisco Unified SRST support and enter call-manager-fallback configuration mode, use the call-manager-fallback command in global configuration mode. To disable Cisco Unified SRST support, use the no form of this command.

call-manager-fallback

no call-manager-fallback

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values.

Command Modes


Global configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.1(5)YD

Cisco SRST 1.0

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(2)XT

Cisco SRST 2.0

This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.

12.2(8)T

Cisco SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco SRST 2.1

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.

Examples

The following example shows how to enter call-manager-fallback configuration mode:


Router(config)# call-manager-fallback

The resulting router prompt is Router(config-cm-fallback)# .

clear voice moh-group statistics

To clear the display of MOH subsystem statistics information and reset the packet counters, use the clear voice moh-group statistics command in privileged EXEC mode.

clear voice moh-group statistics

Syntax Description

This command has no arguments or keywords

Command Modes


Privileged EXEC (#)

Command History

Cisco IOS Release

yCisco Product

Modification

15.0(1)XA

Cisco Unified CME 8.0 Cisco Unified SRST 8.0

This command was introduced.

15.1(1)T

Cisco Unified CME 8.0 Cisco Unified SRST 8.0

This command was integrated into Cisco IOS Release 15.1(1)T.

Usage Guidelines

Use this command to clear the display of MOH subsystem statistics information displayed by the show voice moh-group statistics command.

We recommend that the clear voice moh-group statistics should be used once every two years to reset the packet counters. Each packet counter is of 32 bit size limit and the largest count a packet counter can hold is 4294967296 intervals. This means that with 20 milliseconds packet interval (for G.711), the counters will restart from 0 any time after 2.72 years (2 years and 8 months).

Examples


Router# clear voice moh-group statistics 
All moh group stats are cleared

codec g722-64k

To specify that the G.722 codec should be used for Cisco Unified Survivable Remote Site Telephony (SRST) mode, use the codec g722-64k command in call-manager-fallback configuration mode. To disable this command and restore G.711 as the supported codec for SRST mode, use the no form of this command.

codec g722-64k

no codec g722-64k

Syntax Description

This command has no arguments or keywords.

Command Default

If the codec g722-64k command is not enabled, the G.711 codec is the default for SRST mode.

Command Modes


Call-manager-fallback configuration (config-cm-fallback)

Command History

Release

Cisco Products

Modification

15.0(1)M

Cisco Unified SRST 8.0 Cisco Unified SIP SRST 8.0

This command was introduced.

Usage Guidelines

The G.722 codec should be used for the SRST codec provided the phone supports that codec capability. For phones that do not support G.722 codec, the phones will fall back to the G.711 codec.

Examples

The following example shows how to enable support for the G.722 codec for SRST mode:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# codec g722-64k

conference max-length

To allow Cisco Unified SRST conferencing only if the number of dialed digits are within the maximum length limit, use the conference max-length command. To remove the configuration, use the no form of this command.

conference max-length <value>

no conference max-length

Syntax Description

value

Maximum number of digits that can be dialed. The range is from 3 to 16.

Command Default

No default value is defined for conferencing.

Command Modes

voice register pool (config-register-pool)

Command History

Cisco IOS Release Cisco Product Modification
15.4(3)M Cisco Unified Enhanced SRST 10.5

This command was introduced.

Usage Guidelines

Use the conference max-length command to configure, the Cisco Unified SRST to allow conferencing, only if the dialed digits are within the maximum length limit.

Examples

The following example shows how to configure the maximum number of digits that can be dialed to make a conference call:

Router(config)# voice register pool 1
Router(config-register-pool)# conference max-length 4

conference-pattern blocked

To prevent extensions on an ephone or a voice register pool from initiating a conference to external numbers, use the conference-pattern blocked command. Note that the conference-pattern blocked command does not impact call transfer functions. To remove the configuration, use the no form of this command.

conference-pattern blocked

no conference-pattern blocked

Syntax Description

This command has no arguments or keywords.

Command Default

No default behavior or values.

Command Modes

Ephone configuration (config-ephone)

Voice register pool configuration (config-register-pool)

Command History

Cisco IOS Release Cisco Product Modification
15.4(3)M Cisco Unified Enhanced SRST 10.5

This command was introduced.

Usage Guidelines

Use theconference-pattern blocked command to prevent specific extensions from making conference calls to patterns generally allowed through the transfer-pattern command.

Examples

This example shows how this command prevents extensions from making conference calls to patterns using the transfer-pattern command.

Router(config)# voice regsiter pool 1
Router(config-registetr-pool)# conference-pattern blocked

conference transfer-pattern (call-manager-fallback)

To apply transfer patterns to a conference call using conference softkeys or feature buttons in Cisco Unified SRST, use the conference transfer-pattern command in call-manager-fallback configuration mode. To disable the transfer patterns, use the no form of this command.

conference transfer-pattern

no conference transfer-pattern

Syntax Description

This command has no arguments or keywords.

Command Default

Transfer patterns do not apply to conference calls.

Command Modes

Call-manager-fallback configuration (config-cm-fallback)

Command History

Release

Modification

15.3(2)T

This command was introduced.

Usage Guidelines

There is no check for the conference numbers for call conferencing. To apply transfer patterns for call conferencing, use the conference transfer-pattern command.

When both the transfer-pattern and conference transfer-pattern commands are configured and dialed digits match the configured transfer pattern, conference calls are allowed. However, when the dialed digits do not match any of the configured transfer pattern, the conference call is blocked.

Examples

The following example applies transfer patterns to conference calls:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# transfer-pattern 1234
Router(config-cm-fallback)# conference transfer-pattern

cor (call-manager-fallback)

To configure a class of restriction (COR) on the dial peers associated with directory numbers, use the cor command in call-manager-fallback configuration mode. To disable a COR associated with directory numbers, use the no form of this command.

cor {incoming | outgoing} cor-list-name [cor-list-number starting-number-ending-number | default]

no cor cor-list-name cor-list-number

Syntax Description

incoming

COR list to be used by incoming dial peers.

outgoing

COR list to be used by outgoing dial peers.

cor-list-name

COR list name.

cor-list-number

COR list identifier. The maximum number of COR lists that can be created is 20, comprised of incoming or outgoing dial peers. The first six COR lists are applied to a range of directory numbers. The directory numbers that do not have a COR configuration are assigned to the default COR list, provided that a default COR list has been defined.

starting-number - ending-number

Directory number range; for example, 2000 - 2025.

default

Instructs the COR list to assume behavior according to a predefined default COR list.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(2)XT

Cisco SRST 2.0

This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(8)T

Cisco SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco SRST 2.01

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers. The default keyword was added.

12.3(11)T

Cisco SRST 3.2

The maximum number of COR lists that can be created was increased to 20.

Usage Guidelines

The cor command sets the dial-peer COR parameter for dial peers associated with the directory numbers created during Cisco Unified Communications Manager fallback. A list-based mechanism is provided to assign COR to specific sets of directory numbers during Cisco Unified Communications Manager fallback. The COR functionality provides the ability to deny certain call attempts on the basis of the incoming and outgoing class of restrictions provisioned on the dial peers. This functionality provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers), and applies different restrictions to call attempts from different originators.

COR is used to specify which incoming dial peer can use which outgoing dial peer to make a call. Each dial peer can be provisioned with an incoming and an outgoing COR list.

A default COR is assigned to the directory numbers that do not match any COR list number or number range. The assigned COR is invoked for the dial peers automatically created for each directory number during Cisco Unified Communications Manager fallback registration.

You can have up to 20 COR lists for each incoming and outgoing call. A default COR is assigned to directory numbers that do not match any COR list numbers or number ranges. An assigned COR is invoked for the dial peers and created for each directory number automatically during Communications Manager fallback registration.

If a COR is applied on an incoming dial peer (for incoming calls) and it is a superset or is equal to the COR applied to the outgoing dial peer (for outgoing calls), the call will go through. Voice ports determine whether a call is considered to be incoming or outgoing. If you hook up a phone to an FXS port on a Cisco Unified SRST router and try to make a call from that phone, the call will be considered an incoming call to the router and voice port. If you make a call to the FXS phone, the call will be considered an outgoing call.

By default, an incoming call leg has the highest COR priority. The outgoing COR list has the lowest. If there is no COR configuration for incoming calls on a dial peer, you can make a call from a phone attached to the dial peer, so that the call will go out of any dial peer regardless of the COR configuration on that dial peer. Incoming and outgoing lists are shown in Table 1.

Table 1. Combinations of COR List and Results

COR List on Incoming Dial Peer

COR List on Outgoing Dial Peer

Result

No COR

No COR

Call will succeed.

No COR

COR list applied for outgoing calls

Call will succeed. By default, the incoming dial peer has the highest COR priority when no COR is applied. If you apply no COR for an incoming call leg to a dial peer, the dial peer can make a call out of any other dial peer regardless of the COR configuration on the outgoing dial peer.

COR list applied for incoming calls

No COR

Call will succeed. By default, the outgoing dial peer has the lowest priority. Because there are some COR configurations for incoming calls on the incoming or originating dial peer, it is a superset of the outgoing call’s COR configuration for the outgoing or terminating dial peer.

COR list applied for incoming calls (superset of COR list applied for outgoing calls on the outgoing dial peer)

COR list applied for outgoing calls (subsets of COR list applied for incoming calls on the incoming dial peer)

Call will succeed. The COR list for incoming calls on the incoming dial peer is a superset of the COR list for outgoing calls on the outgoing dial peer.

COR list applied for incoming calls (subset of COR list applied for outgoing calls on the outgoing dial peer)

COR list applied for outgoing calls (supersets of COR list applied for incoming calls on the incoming dial peer)

Call will not succeed. The COR list for incoming calls on the incoming dial peer is not a superset of the COR list for outgoing calls on the outgoing dial peer.

Examples

The following example shows how to set the dial-peer COR parameter for incoming calls to Cisco IP phone dial peers and directory numbers created during Cisco Unified Communications Manager fallback:


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# cor incoming LockforPhoneC 1 5002 – 5010

The following example shows how to set the dial-peer COR parameter for outgoing calls to Cisco IP phone dial peers and directory numbers created during fallback:


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# cor outgoing LockforPhoneC 1 5010 – 5020

The following example shows how to set the dial-peer COR parameter for incoming calls to Cisco IP phone dial peers and directory numbers in the default COR list:


Router(config)# call-manager-fallback 
Router(config-cm-fallback)# cor incoming LockforPhoneC default

The following example shows how sub- and super-COR sets are created. First, a custom dial-peer COR is created with names declared under it:


Router(config)# dial-peer cor custom
Router(config-dp-cor)# name 911
Router(config-dp-cor)# name 1800
Router(config-dp-cor)# name 1900
Router(config-dp-cor)# name local_call

In the following configuration examples, COR lists are created and applied to the dial peer.


Router(config)# dial-peer cor list call911
Router(config-dp-corlist)# member 911
Router(config)# dial-peer cor list call1800
Router(config-dp-corlist)# member 1800
Router(config)# dial-peer cor list call1900
Router(config-dp-corlist)# member 1900
Router(config)# dial-peer cor list calllocal
Router(config-dp-corlist)# member local_call
Router(config)# dial-peer cor list engineering
Router(config-dp-corlist)# member 911
Router(config-dp-corlist)# member local_call
Router(config)# dial-peer cor list manager
Router(config-dp-corlist)# member 911
Router(config-dp-corlist)# member 1800
Router(config-dp-corlist)# member 1900
Router(config-dp-corlist)# member local_call
Router(config)# dial-peer cor list hr
Router(config-dp-corlist)# member 911
Router(config-dp-corlist)# member 1800
Router(config-dp-corlist)# member local_call

In the example below, five dial peers are configured for destination numbers 734...., 1800......., 1900......., 316...., and 911. A COR list is applied to each of the dial peers.


Router(config)# dial-peer voice 1 voip
Router(config-dial-peer)# destination pattern 734....
Router(config-dial-peer)# session target ipv4:1.1.1.1
Router(config-dial-peer)# cor outgoing calllocal
Router(config)# dial-peer voice 2 voip
Router(config-dial-peer)# destination pattern 1800....... 
Router(config-dial-peer)# session target ipv4:1.1.1.1
Router(config-dial-peer)# cor outgoing call1800
Router(config)# dial-peer voice 3 pots 
Router(config-dial-peer)# destination pattern 1900.......
Router(config-dial-peer)# port 1/0/0
Router(config-dial-peer)# cor outgoing call1900
Router(config)# dial-peer voice 4 pots
Router(config-dial-peer)# destination pattern 911
Router(config-dial-peer)# port 1/0/1
Router(config-dial-peer)# cor outgoing call911
Router(config)# dial-peer voice 5 pots
Router(config-dial-peer)# destination pattern 316....
Router(config-dial-peer)# port 1/1/0                 
! No cor is applied.

Finally, the COR list is applied to the individual phone numbers.


Router(config)# call-manager-fallback
Router(config-cm-fallback)# max-conferences 8
Router(config-cm-fallback)# cor incoming engineering 1 1001 - 1001
Router(config-cm-fallback)# cor incoming hr 2 1002 - 1002
Router(config-cm-fallback)# cor incoming manager 3 1003 - 1008

The example configuration allows for the following:

  • Extension 1001 to call 408... numbers, 911 and 316....
  • Extension 1002 to call 408..., 1800 numbers, 911 and 316....
  • Extension 1003 through 1008 to call all of the possible Cisco Unified SRST router numbers
  • All extensions to call 316....

cor (voice register pool)

To configure a class of restriction (COR) on the VoIP dial peers associated with directory numbers, use the cor command in voice register pool configuration mode. To disable a COR associated with directory numbers, use the no form of this command.

cor {incoming | outgoing } cor-list-name {cor-list-number starting-number [- ending-number] | default }

no cor {incoming | outgoing}cor-list-name cor-list-name {cor-list-number starting-number [- ending-number] | default}

Syntax Description

incoming

COR list to be used by incoming dial peers.

outgoing

COR list to be used by outgoing dial peers.

cor-list-name

COR list name.

cor-list-number

COR list identifier.

starting-number

Start of a directory number range, if an ending number is included. Can also be a standalone number.

-

(Optional) Indicator that a full range is configured.

ending-number

(Optional) End of a directory number range.

default

Instructs the COR list to assume behavior according to a predefined default COR list.

Command Default

None

Command Modes


Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(15)ZJ

Cisco SIP SRST 3.0

This command was introduced.

12.3(4)T

Cisco SIP SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was added to Cisco Communications Manager Express (Cisco CME).

Usage Guidelines

The cor command sets the dial-peer COR parameter for dynamically created VoIP dial peers. A list-based mechanism assigns COR parameters to specific set of number ranges. The COR functionality provides the ability to deny certain call attempts on the basis of the incoming and outgoing class of restrictions provisioned on the dial peers. This functionality provides flexibility in network design, allows users to block calls (for example, calls to 900 numbers), and applies different restrictions to call attempts from different originators.

COR specifies which incoming dial peer can use which outgoing dial peer to make a call. Each dial peer can be provisioned with an incoming and an outgoing COR list.

A default COR is assigned to the directory numbers that do not match any COR list number or number range. During Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) registration, a dial peer is created and that dial peer includes a default COR value. The cor command allows you to change the automatically selected default.

In dial-peer configuration mode, build your COR list and add members. Then in voice register pool configuration mode, use the cor command to apply the name of the dial-peer COR list.

You can have up to four COR lists for the Cisco Unified SIP SRST configuration, comprised of incoming or outgoing dial peers. The first four COR lists are applied to a range of phone numbers. The phone numbers that do not have a COR list configuration are assigned to the default COR list, providing that a default COR list has been defined.


Note


Configure the id (voice register pool) command before any other voice register pool commands, including the cor command. The id command identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.

Examples

The following is sample output from the show running-config command:


..
voice register pool 1
 id mac 0030.94C2.A22A
 preference 5
 cor incoming call91 1 91011
 translate-outgoing called 1
 proxy 10.2.161.187 preference 1 monitor probe icmp-ping
 alias 1 94... to 91011 preference 8
 voice-class codec 1
.
.
.
dial-peer cor custom
 name 95
 name 94
 name 91
!
dial-peer cor list call91
 member 91
!
dial-peer voice 91500 pots
 corlist incoming call91
 corlist outgoing call91
 destination-pattern 91500
 port 1/0/0
.
.
.

credentials

To enter credentials configuration mode to configure a certificate for a Cisco Unified CME CTL provider or for Cisco Unified SRST router communication to Cisco Unified Communications Manager, use the credentials command in global configuration mode. To set all commands in credentials configuration mode to the default of nonsecure, use the no form of this command.

credentials

no credentials

Syntax Description

This command has no arguments or keywords.

Command Default

Credentials are not provided.

Command Modes


Global configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.3(14)T

Cisco SRST 3.3

This command was introduced for Cisco Unified SRST.

12.4(4)XC

Cisco Unified CME 4.0

This command was introduced for Cisco Unified CME.

Cisco IOS XE Fuji Release, 16.7.1

Unified SRST 12.1

This command was introduced for Unified SRST support on Cisco 4000 Series Integrated Services Router.

Cisco IOS XE Dublin 17.10.1a

Cisco Unified SRST 14.3

Introduced support for YANG models.

Usage Guidelines

This command is used to configure credentials service for Cisco Unified CME and Cisco Unified SRST.

Cisco Unified CME

This command is used with Cisco Unified CME phone authentication to configure a CTL provider on each Cisco Unified CME router on which the CTL client is not running. That is, if there is a primary and a secondary Cisco Unified CME router and the CTL client is running on the primary router, a CTL provider must be configured on the secondary router, and vice versa. If the CTL client is running on a router that is not a Cisco Unified CME router, CTL providers must be configured on all Cisco Unified CME routers.

Credentials service for Cisco Unified CME runs on default port 2444.

Cisco Unified SRST

The credential server provides certificates to any device that requests a certificate. The credentials server does not request any data from a client; thus no authentication is necessary. When the client, Cisco Unified Communications Manager, requests a certificate, the credentials server provides the certificate. Cisco Unified Communications Manager exports the certificate to the phone, and the Cisco Unified IP phone holds the SRST router certificate in its configuration file. The device certificate for secure SRST routers is placed in the configuration file of the Cisco Unified IP phone because the entry limit in the certificate trust list (CTL) of Cisco Unified Communications Manager is 32.

Credentials service for SRST runs on default port 2445. Cisco Unified Communications Manager connects to port 2445 on the secure SRST router and retrieves the secure SRST device certificate during the TLS handshake.

Activate this command on all SRST routers.


Caution


For security reasons, credentials service should be deactivated on all SRST routers after provisioning to Cisco Unified Communications Manager is completed.


Examples

Examples

The following example configures a CTL provider on the Cisco Unified CME router with the IP address 172.19.245.1. CTL providers must be configured on all Cisco Unified CME routers on which the CTL client is not running.


Router(config)# credentials
Router(config-credentials)# ip source-address 172.19.245.1 port 2444
Router(config-credentials)# trustpoint cmeca
Router(config-credentials)# ctl-service admin user4 secret 0 c89L8o

Examples

The following example enters credentials configuration mode and sets the IP source address and the trustpoint:


Router(config)# credentials 
Router(config-credentials)# ip source-address 10.6.21.4 port 2445
Router(config-credentials)# 
trustpoint srstca

date-format (call-manager-fallback)

To set the date display format on all the Cisco IP phones attached to the router, use the date-format command in call-manager-fallback configuration mode. To disable the date display format, use the no form of this command.

date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd}

no date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd}

Syntax Description

mm-dd-yy

Sets the date format to month, day, year. Each element requires a two-digit number. This format is the default setting.

dd-mm-yy

Sets the date format to day, month, year. Each element requires a two-digit number.

yy-dd-mm

Sets the date format to year, day, month. Each element requires a two-digit number.

yy-mm-dd

Sets the date format to year, month, day. Each element requires a two-digit number.

Command Default

mm-dd-yy

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.2(2)XT

Cisco SRST 2.0

This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(8)T

Cisco SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco SRST 2.01

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.

12.2(15)ZJ

Cisco SRST 3.0

This command was integrated into Cisco IOS Release 12.2(15)ZJ.

12.3(4)T

Cisco SRST 3.0

This command was integrated into Cisco IOS Release 12.3(4)T.

Examples

The following example sets the date format:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# date-format mm-dd-yy

default-destination

To create a default call routing path for incoming calls on Foreign Exchange Office (FXO) ports during a WAN outage, use the default-destination command in call-manager-fallback configuration mode. To delete the default destination number on the Cisco Unified SRST router, use the no form of this command.

default-destination telephone-number

no default-destination telephone-number

Syntax Description

telephone-number

Telephone number of the default destination.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.1(5)YD

Cisco SRST 1.0

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.

12.2(2)XT

Cisco SRST 2.0

This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.

12.2(8)T

Cisco SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.

12.2(8)T1

Cisco SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco SRST 2.01

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.

Usage Guidelines

The default-destination command creates a temporary Private Line Auto Ringdown (PLAR) connection configuration on FXO ports during fallback. The default-destination command has been obsoleted by the alias command, which applies to all port types. Use of the alias command is recommended over the default-destination command.

Examples

The following example sets the default destination to 40802:


Router(config)# call-manager-fallback
Router(config-cm-fallback)# default-destination 40802

description (voice moh-group)

To display a brief description specific to a MOH group, use the description command in voice moh-group configuration mode. To remove the description, use the no form of this command.

description string

no description

Syntax Description

string

An alphanumeric string to add a brief description specific to a MOH group. Maximum length: 80 characters including spaces.

Command Default

No MOH group description is configured.

Command Modes


V
oice moh-group configuration (config-voice-moh-group)

Command History

Cisco IOS Release

Cisco Product

Modification

15.0(1)XA

Cisco Unified CME 8.0 Cisco Unified SRST 8.0

This command was introduced

15.1(1)T

Cisco Unified CME 8.0 Cisco Unified SRST 8.0

This command was integrated into Cisco IOS Release 15.1(1)T.

Usage Guidelines

This command allows you to type a brief text describing a specific voice-moh-group. You can use maximum 80 characters, including spaces to describe a MOH group.

Examples

The following example provides a description for voice-moh-group1:


Router(config)# 
Router(config-voice-moh-group)#
Router(config-voice-moh-group) descri
ption this is a moh group for sales
 

device-id

To specify the device ID of a phone type, use the device-id command in ephone-type configuration mode. To reset to the default value, use the no form of this command.

device-id number

no device-id

Syntax Description

number

Device ID of the phone type. Range: 1 to 2,147,483,647. Default: 0. SeeTable 1 for a list of supported device IDs.

Command Default

Device ID is 0.

Command Modes


Ephone-type configuration (config-ephone-type)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)XZ

Cisco Unified CME 4.3 Cisco Unified SRST 4.3

This command was introduced.

Usage Guidelines

This command specifies the device ID of the type of phone being added with the ephone-type template. If this command is set to the default value of 0, the ephone-type is invalid.

Table 2. Supported Values for Ephone-Type Commands

Supported Device

device-id

num-buttons

max-presentation

Cisco Unified IP Conference Station 7937G

431

1

6

Nokia E61

376

1

1

Examples

The following example shows the device ID is set to 376 for the Nokia E61 when creating the ephone-type template:


Router(config)# ephone-type E61
Router(config-ephone-type)# device-id 376
Router(config-ephone-type)# device-name E61 Mobile Phone

device-name

To assign a name to a phone type in an ephone-type template, use the device-name command in ephone-type configuration mode. To remove the name, use the no form of this command.

device-name name

no device-name

Syntax Description

name

String that identifies this phone type. Value is any alphanumeric string up to 32 characters.

Command Default

No name is assigned to this phone type.

Command Modes


Ephone-type configuration (config-ephone-type)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)XZ

Cisco Unified CME 4.3 Cisco Unified SRST 4.3

This command was introduced.

Usage Guidelines

This command specifies a device name for the type of phone being added with the ephone-type template.

Examples

The following example shows that the name “E61 Mobile Phone” is assigned to a phone type when creating the ephone-type template:


Router(config)# ephone-type E61
Router(config-ephone-type)# device-id 376
Router(config-ephone-type)# device-name E61 Mobile Phone

device-type

To specify the phone type, use the device-type command in ephone-type configuration mode. To reset to the default value, use the no form of this command.

device-type phone-type

no device-type

Syntax Description

phone-type

Device type of the phone. See Table 1 for a list of supported device types. Default value is the same value entered with the ephone-type command.

Command Default

Device type is the same value that is entered with the ephone-type command.

Command Modes


Ephone-type configuration (config-ephone-type)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)XZ

Cisco Unified CME 4.3 Cisco Unified SRST 4.3

This command was introduced.

12.4(20)T

Cisco Unified CME 7.0 Cisco Unified SRST 7.0

This command was integerated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

This command specifies the device type of the phone being added with the ephone-type template. The device type is set to the same value as the ephone-type command unless you use this command to change the value.

This command must be set to one of the following supported values.

Table 3. Supported Values for Ephone-Type Commands

Supported Device

device-id

device-type

num-buttons

max-presentation

Cisco Unified IP Phone 7915 Expansion Module with 12 buttons

227

7915

12

0 (default)

Cisco Unified IP Phone 7915 Expansion Module with 24 buttons

228

7915

24

0

Cisco Unified IP Phone 7916 Expansion Module with 12 buttons

229

7916

12

0

Cisco Unified IP Phone 7916 Expansion Module with 24 buttons

230

7916

24

0

Cisco Unified IP Conference Station 7937G

431

7937

1

6

Cisco Unified IP Phone 8941

586

8941

4

3

Cisco Unified IP Phone 8945

585

8945

4

3

Nokia E61

376

E61

1

1

Examples

The following example shows the device type set to 7915 in the ephone-type template for the Cisco Unified IP Phone 7915 Expansion Module with 12 buttons:


Router(config)# ephone-type 7915-12 addon
Router(config-ephone-type)# device-id 227
Router(config-ephone-type)# device-name 7915-12
Router(config-ephone-type)# device-type 7915

dialplan-pattern (call-manager-fallback)

To create a global prefix that can be used to expand the extension numbers of inbound and outbound calls into fully qualified E.164 numbers, use the dialplan-pattern command in call-manager-fallback configuration mode. To disable the dialplan-pattern command settings, use the no form of this command.

dialplan-pattern tag pattern extension-length extension-length [extension-pattern extension-pattern] [no-reg]

no dialplan-pattern tag [pattern extension-length extension-length extension-pattern extension-pattern] [no-reg]

Syntax Description

tag

Dial-plan string tag used before a ten-digit telephone number. The tag number is from 1 to 5.

pattern

Dial-plan pattern, such as the area code, the prefix, and the first one or two digits of the extension number, plus wildcard markers or dots (.) for the remainder of the extension number digits.

extension-length

Sets the number of extension digits that will appear as a caller ID.

extension-length

The number of extension digits. The extension length must match the setting for IP phones in Cisco Unified Communications Manager mode. The range is from 1 to 32.

extension-pattern

(Optional) Sets an extension number’s leading digit pattern when it is different from the E.164 telephone number’s leading digits defined in the pattern variable.

extension-pattern

(Optional) The extension number’s leading digit pattern. Consists of one or more digits and wildcard markers or dots (.). For example, 5.. would include extensions 500 to 599; 5... would include extensions 5000 to 5999. The extension pattern must match the setting for IP phones in Cisco Unified Communications Manager mode.

no-reg

(Optional) Prevents the E.164 numbers in the dial peer from registering with the gatekeeper.

Command Default

No default behavior or values.

Command Modes


Call-manager-fallback configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.1(5)YD

Cisco SRST 1.0

This command was introduced on the Cisco 2600 series and Cisco 3600 series multiservice routers and on the Cisco IAD2420 series.

12.2(2)XT

Cisco SRST 2.0

This command was implemented on the Cisco 1750 and Cisco 1751 multiservice routers.

12.2(8)T

Cisco SRST 2.0

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.

12.2(8)T1

Cisco SRST 2.0

This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.

12.2(11)T

Cisco SRST 2.01

This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.

12.2(11)YT

Cisco SRST 2.1

The extension-pattern keyword was added.

12.4(20)T

Cisco Unified CME 7.0 Cisco Unified SRST 7.0

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

The dialplan-pattern command builds additional dial peers. For example, if a hidden POTS dial peer is created, such as the following:


Router(config)# dial-peer voice 20001 pots
Router(config-dial-peer)# destination-pattern 1001
Router(config-dial-peer)# voice-port 50/0/2

and a dial-plan pattern is created, such as 40855510.., then an additional dial peer will be created that allows calls to both the 1001 and 4085551001 numbers. For example:


Router(config)# dial-peer voice 20002 pots
Router(config-dial-peer)# destination-pattern 4085551001
Router(config-dial-peer)# voice-port 50/0/2

Both dial peers can be seen with the show dial-peer voice command.

The dialplan-pattern command also creates a global prefix that can be used by inbound calls (calls to an IP phone in a Cisco Unified SRST system) and outbound calls (calls made from an IP phone in a Cisco Unified SRST system) to expand their extension numbers to fully qualified E.164 numbers.

For inbound calls (calls to an IP phone in a Cisco Unified SRST system) where the calling party number matches the dial-plan pattern, the call is considered a local call and has a distinctive ring that identifies the call as internal. Any calling party number that does not match the dial-plan pattern is considered an external call and has a distinctive ring that is different from the internal ringing.

For outbound calls, the dialplan-pattern command converts the calling party’s extension number to an E.164 calling party number. Outbound calls that do not use an E.164 number and go through a PRI connection to the PSTN may be rejected by the PRI link as the calling party identifier.

If there are multiple patterns, called-party numbers are checked in numeric order, starting with pattern 1, until a match is found or until the last pattern has been checked. The valid dial-plan pattern with the lowest tag is used as a prefix to all local Cisco IP phones.

When extension-pattern extension-pattern keyword and argument are used, the leading digits of an extension pattern are stripped and replaced with the corresponding leading digits of the dial plan. For example, the following command maps all extension numbers 4xx to the PSTN number 40855501xx, so that extension 412 corresponds to 4085550112.


Router(config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 4085550100 extension-length 3 extension-pattern 4.. 

The number of extension-pattern argument characters must match the number set for the extension-length argument. For example, if the extension-length is 3, the extension-pattern can be 8.., 1.., 51., and so forth.

A dial-plan pattern is required to register the Cisco IP phone lines with a gatekeeper. The no-reg keyword provides the option of not registering specific numbers to the gatekeeper so that those numbers can be used for other telephony services.

Examples

The following example shows how to create dial-plan pattern 1 for extension numbers 5000 to 5099 with a prefix of 408555. If an inbound calling party number (4085555044) matches dial-plan pattern 1, the recipient phone will display an extension (5044) as the caller ID and use an internal ringing tone. If an outbound calling party extension number (5044) matches dial-plan pattern 1, the calling party extension will be converted to an E.164 number (4085555044). The E.164 calling party number will appear as the caller ID.


Router(config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 40855550.. extension-length 4 extension-pattern 50..

In the following example, the dialplan-pattern command creates dial-plan pattern 1 for extensions 800 to 899 with the telephone prefix starting with 4085559. As each number in the extension pattern is declared with the number command, two POTs dial peers are created. In the example, they are 801 (an internal office number) and 4085559001 (an external number).


Router(config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 40855590.. extension-length 3 extension-pattern 8..

The following example shows a configuration for two Cisco Unified SRST systems. Each is configured with the same dialplan-pattern commands, but one system uses 50.. and the other uses 60.. for extension numbers. Calls from the “50..” system to the “60..” system, and vice versa, are treated as internal calls. Calls that go across an H.323 network and calls that go to a PSTN through an ISDN interface on one of the configured Cisco Unified SRST routers are represented as E.164.


Router(config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 40855550.. extension-length 4 extension-pattern 50..
Router(config-cm-fallback)# dialplan-pattern 2 51055560.. extension-length 4 extension-pattern 60..

digit collect kpml

To enable Key Press Markup Language (KPML) digit collection on a SIP phone, use the digit collect kpml command in voice register pool or voice register template configuration mode. To disable KPML, use the no form of this command.

digit collect kpml

no digit collect kpml

Syntax Description

This command has no arguments or keywords.

Command Default

KPML digit collection is enabled.

Command Modes


Voice register pool configuration
Voice register template configuration

Command History

Release

Cisco Product

Modification

12.4(11)XJ

Cisco Unified CME 4.1 Cisco Unified SRST 4.1

This command was introduced.

12.4(15)T

Cisco Unified CME 4.1 Cisco Unified SRST 4.1

This command was integrated into Cisco IOS Release 12.4(15)T.

Usage Guidelines

KPML is enabled by default for all directory numbers on the phone. A dial plan assigned to a phone has priority over KPML. Use the no digit collect kpml command to disable KPML on a phone.

If you use a voice register template to apply a command to a phone and you also use the same command in voice register pool configuration mode for the same phone, the value that you set in voice register pool configuration mode has priority.

KPML is not supported on the Cisco IP Phone 7905, 7912, 7940, or 7960.

Examples

The following example shows KPML enabled on SIP phone 4:


Router(config)# voice register pool 4
Router(config-register-pool)# digit collect kpml

dtmf-relay (voice register pool)

To specify the list of DTMF relay methods that can be used to relay dual-tone multifrequency (DTMF) audio tones between Session Initiation Protocol (SIP) endpoints, use the dtmf-relay command in voice register pool configuration mode. To send the DTMF audio tones as part of an audio stream, use the no form of this command.

dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]

no dtmf-relay

Syntax Description

cisco-rtp

Forwards DTMF audio tones by using Real-Time Transport Protocol (RTP) with a Cisco proprietary payload type. This keyword is supported only for dial peers that are created by incoming REGISTERs from a SIP gateway. It is not supported for dial peers that are created by a SIP Cisco IP phone.

rtp-nte

Forwards DTMF audio tones by using Real-Time Transport Protocol (RTP) with a Named Telephone Event (NTE) payload.

sip-notify

Forwards DTMF audio tones by using SIP-NOTIFY messages. This keyword is supported only for dial peers that are created by incoming REGISTERs from a SIP gateway. It is not supported for dial peers that are created by a SIP Cisco IP phone.

Command Default

DTMF tones are disabled and sent in-band. That is, they remain in the audio stream.

Command Modes


Voice register pool configuration

Command History

Cisco IOS Release

Cisco Product

Modification

12.3(4)T

Cisco SIP SRST 3.0

This command was introduced.

12.4(4)T

Cisco CME 3.4 Cisco SIP SRST 3.4

This command was added to Cisco Unified CME.

Cisco IOS XE Amsterdam 17.2.1r

Cisco Unified SIP SRST 12.8

Introduced support for YANG models.

Usage Guidelines

During Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) or Cisco Unified CME registration, a dial peer is created and that dial peer has a default DTMF relay of in-band.

This command command allows you to change the default to a desired value. You must use one or more keywords when configuring this command.

DTMF audio tones are generated when you press a button on a Touch-Tone phone. The tones are compressed at one end of the call and when the digits are decompressed at the other end, there is a risk that they can become distorted. DTMF relay reliably transports the DTMF audio tones generated after call establishment out-of-band.

The SIP Notify method sends Notify messages bidirectionally between the originating and terminating gateways for a DTMF event during a call. If multiple DTMF relay mechanisms are enabled on a SIP dial peer and are negotiated successfully, the SIP Notify method takes precedence.

SIP Notify messages are advertised in an Invite message to the remote end only if the dtmf-relay command is set.

For SIP calls, the most appropriate methods to transport DTMF tones are RTP-NTE or SIP-NOTIFY.


Note


The cisco-rtp keyword is a proprietary Cisco implementation. If the proprietary Cisco implementation is not supported, the DTMF relay feature does not function, and the gateway sends DTMF tones in-band.
  • The sip-notify keyword is available only if the VoIP dial peer is configured for SIP.

Examples

Examples

The following example shows how to enable the RTP-NTE and SIP-NOTIFY mechanisms for DTMF relay for SIP phone 4:


Router(config)# voice register pool 4
Router(config-register-pool)# dtmf-relay rtp-nte sip-notify 

Examples

The following is sample output from the show running-config command that shows that voice register pool 1 has been set up to send DTMF tones:


voice register pool 1 
 application SIP.app
 incoming called-number 308
 voice-class codec 1
 dtmf-relay rtp-nte

elin

To create a PSTN number that replaces a 911 caller’s extension, use the elin command in voice emergency response location configuration mode. To remove the number, use the no form of this command.

elin [1 | 2] number

no elin [1 | 2]

Syntax Description

number

PSTN number that replaces a 911 caller’s extension.

Command Default

No replacement number is created.

Command Modes


Voice emergency response location configuration (cfg-emrgncy-resp-location)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)T

Cisco Unified CME 4.1 Cisco Unified SRST 4.1 Cisco Unified SIP SRST 4.1

This command was introduced. For Cisco Unified CME, this command is supported in SRST fallback mode only.

12.4(15)XY

Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)

This command was added for Cisco Unified CME.

12.4(20)T

Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

Use this command to specify the ELIN, a PSTN number that will replace the caller’s extension. The PSAP will see this number and use it to query the ALI database to locate the caller. It is also used by the PSAP for callbacks. You can optionally configure a second ELIN using the elin 2 command. If two ELINs are configured, the system selects an ELIN using a round-robin algorithm. If an ELIN is not defined for the ERL, the PSAP will see the original calling number.

Examples

In the following example, all IP phones with the IP address of 10.X.X.X or 192.168.X.X are automatically associated with this ERL. If one of the phones dials 911, its extension is replaced with 408 555-0100 before it goes to the PSAP. The PSAP will see that the caller’s number is 408 555-0100.


voice emergency response location 1
 elin 1 4085550100
 subnet 10.0.0.0 255.0.0.0
 subnet 2 192.168.0.0 255.255.0.0

elin (voice emergency response settings)

To create a default ELIN that is used if no ERL has a subnet mask that matches the current 911 caller’s IP phone address, use the elin command in voice emergency response settings configuration mode. To remove the number, use the no form of this command.

elin number

no elin

Syntax Description

number

An E.164 number to be used as the default ELIN.

Command Default

No default ELIN number is created.

Command Modes


Voice emergency response settings configuration (cfg-emrgncy-resp-settings)

Command History

Cisco IOS Release

Cisco Product

Modification

12.4(15)XY

Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)

This command was introduced.

12.4(20)T

Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0

This command was integrated into Cisco IOS Release 12.4(20)T.

Usage Guidelines

This command specifies an E.164 number to be the default ELIN if the 911 caller’s IP phone address does not match the subnet of any location in any ERL zone. The default ELIN can be an existing ELIN already defined in an ERL or it can be unique.

Examples

In this example, the ELIN (4085550101) defined in the voice emergency response settings configuration is used if the 911 caller’s IP phone address does not match any of the voice emergency response locations. After the 911 call is placed to the PSAP, the PSAP has 120 minutes to call back 408 555-0101 to reach the 911 caller. If the call history has expired (after 120 minutes), any callback is routed to extension 7500.


voice emergency response settings
callback 7500
elin 4085550101
expiry 120