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Cisco SRST and SIP SRST Command Reference (All Versions)
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Book-Length PDF: Cisco Unified SRST and Cisco Unified SIP SRST Command Reference (All Versions)
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Cisco Unified SRST and Cisco Unified SIP SRST Command Reference: A through M
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Cisco Unified SRST and Cisco Unified SIP SRST Command Reference: N through Z
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Cisco Unified SRST and Cisco Unified SIP SRST Command Reference: Index
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Table Of Contents
Command Reference: A through M
address (voice emergency response location)
after-hour exempt (voice register pool)
after-hours block pattern (call-manager-fallback)
after-hours date (call-manager-fallback)
after-hours day (call-manager-fallback)
application (call-manager-fallback)
application (voice register global)
application (voice register pool)
authenticate (voice register global)
call-forward busy (call-manager-fallback)
call-forward noan (call-manager-fallback)
call-forward pattern (call-manager-fallback)
clear voice moh-group statistics
date-format (call-manager-fallback)
dialplan-pattern (call-manager-fallback)
dtmf-relay (voice register pool)
elin (voice emergency response settings)
external-ring (voice register global)
h450 h450-2 timeout (voice service voip)
huntstop (call-manager-fallback)
incoming called-number (voice register pool)
ip qos dscp (call-manager-fallback)
ip source-address (call-manager-fallback)
ip source-address (credentials)
keepalive (call-manager-fallback)
limit-dn (call-manager-fallback)
location (voice emergency response zone)
logging (voice emergency response settings)
max-conferences (call-manager-fallback)
max-dn (call-manager-fallback)
max-ephones (call-manager-fallback)
maximum bit-rate (cm-fallback-video)
max-dn (voice register global)
max-pool (voice register global)
max registrations (voice register pool)
moh-live (call-manager-fallback)
multicast moh (call-manager-fallback)
mwi expires (call-manager-fallback)
mwi reg-e164 (call-manager-fallback)
mwi relay (call-manager-fallback)
mwi sip-server (call-manager-fallback)
Command Reference: A through M
Last Updated: April 23, 2012This 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 master 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 E&M 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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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-fallbackRouter(config-cm-fallback)# access-code bri 8 direct-inward-dialThe following example sets the access-code command for E&M 8:Router(config)# call-manager-fallbackRouter(config-cm-fallback)# access-code e&m 8The following example sets the access-code command for FXO 9:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# access-code fxo 9The following example sets the access-code command for PRI 9:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# access-code pri 9 direct-inward-dialRelated Commands
call-manager-fallback
Enables Cisco Unified SRST feature support and enters call-manager-fallback configuration mode.
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
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
Related Commands
device-id
Specifies the device ID for a phone type in an ephone-type template.
device-name
Assigns a name to a phone type in an ephone-type template.
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
Command Default
The civic address is not defined.
Command Modes
Voice emergency response location configuration (cfg-emrgncy-resp-location)
Command History
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 60subnet 1 209.165.200.224 255.255.0.0elin 1 4085550100name Cookies and More Incorporated,address I,408,5550100,,11902,,,Main Street,Emerald City,CA,Idina Menzel,1,,,,,,Related Commands
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
12.4(4)T
Cisco CME 3.4
Cisco SRST 3.4This 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
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 23Router(config-register-pool)# after-hour exemptThe 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 2Router(config-register-pool)# number 5001Router(config-register-pool)# after-hour exemptRelated Commands
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
Command Default
No pattern is defined.
Command Modes
Call-manager-fallback configuration
Command History
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.
Note
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-fallbackRouter(config-cm-fallback)# after-hours block pattern 1 9011Related Commands
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
Command Default
No time period based on date is defined for call blocking.
Command Modes
Call-manager-fallback configuration
Command History
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
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-fallbackRouter(config-cm-fallback)# after-hours date jan 1 00:00 00:00Related Commands
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
Command Default
No time period based on date is defined for call blocking.
Command Modes
Call-manager-fallback configuration
Command History
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
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-fallbackRouter(config-cm-fallback)# after-hours date jan 1 00:00 00:00Related Commands
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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
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 20In 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
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-fallbackRouter(config-cm-fallback)# alias 1 60.. to 5001 preference 2In 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-fallbackRouter(config-cm-fallback)# alias 1 5... to 6000The 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-fallbackRouter(config-cm-fallback)# alias 1 1000 to 1000 preference 1 cfw 2000 timeout 10Router(config-cm-fallback)# max-dn 10 preference 2Router(config-cm-fallback)# call-forward busy 3000Router(config-cm-fallback)# call-forward noan 3000 timeout 10Related Commands
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
Command Default
None
Command Modes
Voice register pool configuration
Command History
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
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 poolRouter(config-register-pool)# alias 1 50.. to 5001Related Commands
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
Command Default
DEFAULT system session application
Command Modes
Call-manager-fallback configuration
Command History
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-fallbackRouter(config-cm-fallback) application app-xferRelated Commands
call-manager-fallback
Enables Cisco Unified SRST and enters call-manager-fallback configuration mode.
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
Command Default
Default application on router
Command Modes
Voice register global configuration
Command History
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This 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
Examples
The following example shows how to set the Tcl IVR application globally for all SIP phones:
Router(config)# voice register globalRouter(config-register-global)# mode cmeRouter(config-register-global)# application sipapp2Related Commands
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
Command Default
Default application on router
Command Modes
Voice register pool configuration
Command History
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
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 1Router(config-register-pool) application sipapp2The 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 1id network 172.16.0.0 mask 255.255.0.0application SIP.appvoice-class codec 1Related Commands
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
Command Default
The default size for attempted registration table is 10.
Command Modes
voice register global
Command History
15.1(2)T
Cisco Unified CME 8.1
Cisco Unified SRST 8.1This 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
!
Related Commands
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
Command Default
Authenticate mode is disabled.
Command Modes
Voice register global configuration
Command History
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 globalRouter(config-register-global)# mode cmeRouter(config-register-global)# authenticate registerRelated Commands
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
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
Examples
The following example shows b2bua included in the configuration for voice dial peer 1:
dial-peer voice 1 voipdestination-pattern 4...session target ipv4:10.5.49.80session protocol sipv2dtmf-relay sip-notifyb2buaRelated Commands
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 configurationCommand History
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This command was introduced.
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
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 4Router(config-register-dn)# number 5001Router(config-register-dn)# call-forward b2bua all 5005The following example shows how to forward all incoming calls for extension 5001 on pool number 4, to extension 5005.
Router(config)# voice register pool 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua all 5005Related Commands
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 configurationCommand History
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This 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
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 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua busy 5005Related Commands
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
Command Default
Disabled (no voice-mail box is selected for call forwarding).
Command Modes
Voice register dn configuration
Voice register pool configurationCommand History
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This 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
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 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua mailbox 5005Related Commands
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
Command Default
Disabled (no incoming calls to an extension that does not answer are forwarded).
Command Modes
Voice register dn configuration
Voice register pool configurationCommand History
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This 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
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 4Router(config-register-pool)# number 5001Router(config-register-pool)# call-forward b2bua noan 5005 timeout 10Related Commands
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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-fallbackRouter(config-cm-fallback)# call-forward busy 5005The 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-fallbackRouter(config-cm-fallback)# call-forward busy 50..
Note
Related Commands
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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-fallbackRouter(config-cm-fallback)# call-forward noan 5005 timeout 10The 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-fallbackRouter(config-cm-fallback)# call-forward noan 50.. timeout 10
Note
Related Commands
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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-fallbackRouter(config-cm-fallback)# call-forward pattern 4...The following example forwards all calls using the H.450.3 standard:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# call-forward pattern .TRelated Commands
call-manager-fallback
Enables Cisco Unified SRST feature support and enters call-manager-fallback configuration mode.
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
Examples
The following example shows how to enter call-manager-fallback configuration mode:
Router(config)# call-manager-fallbackThe resulting router prompt is
Router(config-cm-fallback)#.Related Commands
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
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 statisticsAll moh group stats are clearedRelated Commands
show voice moh-group statistics
Displays the MOH subsystem statistics information
show voice moh-group
Displays the MOH groups configured
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
15.0(1)M
Cisco Unified SRST 8.0
Cisco Unified SIP SRST 8.0This 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-fallbackRouter(config-cm-fallback)# codec g722-64kRelated Commands
codec (voice register pool)
To specify the codec to be used when setting up a call for a SIP phone or group of SIP phones in Cisco Unified CME or Cisco Unified SIP SRST, use the codec command in voice register pool configuration mode. To disable a specified codec, use the no form of this command.
codec codec-type [bytes]
no codec
Syntax Description
Command Default
Default is G.729 8K bps.
Command Modes
Voice register pool configuration (config-register-pool)
Command History
Usage Guidelines
This command changes the automatically selected codec for the dial peer that is dynamically created when the SIP phone registers from the default (G.729) to the specified codec.
This command affects calls between two phones on the same Cisco Unified CME or Cisco Unified SRST router, such as between an IP phone and another IP phone or between an IP phone and an FXS analog phone.
If codec values for the dial peers of a connection do not match, the call fails. For example, the default codec for the POTS dial peer for an SCCP phone is G.711 and the default codec for a VoIP dial peer for a SIP phone is G.729. If neither the SCCP phone nor the SIP phone has been specifically configured to change the codec, calls between the two IP phones on the same router will produce a busy signal caused by the mismatched default codecs. For calls to other phones on the same Cisco router, a SIP phone that is configured to use G.711 will always have its calls set up to use G.711. If the phone participates in a call on a line that is shared with a phone that is configured for G.711, it must use G.711.
For calls to external phones; that is, phones that are not in the same Cisco Unified CME (such as VoIP calls), the codec is negotiated based on the protocol that is used for the call (such as H.323). The Cisco Unified CME plays no part in the negotiation.
This command sets the codec configuration for an individual phone and overrides any previously configured codec selection set with the voice-class codec command or the codec (telephony-service) command.
Note
Examples
The following example shows how to set the codec to g711 for a SIP phone in pool 1:
Router(config)# voice register pool 1
Router(config-register-pool)# codec g711ulaw
.
.
.
The following partial sample from the show voice register pool command shows the configuration for voice register pool 1:
pool tag 1
Config
MAC address is 0012.DABF.26BE
Type is 7960
Number list 1: dn 1Proxy Ip address is 0.0.0.0
Codec is g711ulaw
...
Dialpeers createddial-peer voice 4003 voip
destination-pattern 6667
session target ipv
session protocol sip2v
codec g711ulaw
Related Commands
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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.
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-fallbackRouter(config-cm-fallback)# cor incoming LockforPhoneC 1 5002 - 5010The 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-fallbackRouter(config-cm-fallback)# cor outgoing LockforPhoneC 1 5010 - 5020The 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-fallbackRouter(config-cm-fallback)# cor incoming LockforPhoneC defaultThe 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 customRouter(config-dp-cor)# name 911Router(config-dp-cor)# name 1800Router(config-dp-cor)# name 1900Router(config-dp-cor)# name local_callIn the following configuration examples, COR lists are created and applied to the dial peer.
Router(config)# dial-peer cor list call911Router(config-dp-corlist)# member 911Router(config)# dial-peer cor list call1800Router(config-dp-corlist)# member 1800Router(config)# dial-peer cor list call1900Router(config-dp-corlist)# member 1900Router(config)# dial-peer cor list calllocalRouter(config-dp-corlist)# member local_callRouter(config)# dial-peer cor list engineeringRouter(config-dp-corlist)# member 911Router(config-dp-corlist)# member local_callRouter(config)# dial-peer cor list managerRouter(config-dp-corlist)# member 911Router(config-dp-corlist)# member 1800Router(config-dp-corlist)# member 1900Router(config-dp-corlist)# member local_callRouter(config)# dial-peer cor list hrRouter(config-dp-corlist)# member 911Router(config-dp-corlist)# member 1800Router(config-dp-corlist)# member local_callIn 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 voipRouter(config-dial-peer)# destination pattern 734....Router(config-dial-peer)# session target ipv4:1.1.1.1Router(config-dial-peer)# cor outgoing calllocalRouter(config)# dial-peer voice 2 voipRouter(config-dial-peer)# destination pattern 1800.......Router(config-dial-peer)# session target ipv4:1.1.1.1Router(config-dial-peer)# cor outgoing call1800Router(config)# dial-peer voice 3 potsRouter(config-dial-peer)# destination pattern 1900.......Router(config-dial-peer)# port 1/0/0Router(config-dial-peer)# cor outgoing call1900Router(config)# dial-peer voice 4 potsRouter(config-dial-peer)# destination pattern 911Router(config-dial-peer)# port 1/0/1Router(config-dial-peer)# cor outgoing call911Router(config)# dial-peer voice 5 potsRouter(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-fallbackRouter(config-cm-fallback)# max-conferences 8Router(config-cm-fallback)# cor incoming engineering 1 1001 - 1001Router(config-cm-fallback)# cor incoming hr 2 1002 - 1002Router(config-cm-fallback)# cor incoming manager 3 1003 - 1008The example configuration allows for the following:
•
•
•
•
Related Commands
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-number starting-number [- ending-number] | default}
Syntax Description
Command Default
None
Command Modes
Voice register pool configuration
Command History
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
Examples
The following is sample output from the show running-config command:
..voice register pool 1id mac 0030.94C2.A22Apreference 5cor incoming call91 1 91011translate-outgoing called 1proxy 10.2.161.187 preference 1 monitor probe icmp-pingalias 1 94... to 91011 preference 8voice-class codec 1...dial-peer cor customname 95name 94name 91!dial-peer cor list call91member 91!dial-peer voice 91500 potscorlist incoming call91corlist outgoing call91destination-pattern 91500port 1/0/0...Related Commands
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
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
Examples
Cisco Unified CME
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)# credentialsRouter(config-credentials)# ip source-address 172.19.245.1 port 2444Router(config-credentials)# trustpoint cmecaRouter(config-credentials)# ctl-service admin user4 secret 0 c89L8oCisco Unified SRST
The following example enters credentials configuration mode and sets the IP source address and the trustpoint:
Router(config)# credentialsRouter(config-credentials)# ip source-address 10.6.21.4 port 2445Router(config-credentials)# trustpoint srstcaRelated Commands
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
Command Default
mm-dd-yy
Command Modes
Call-manager-fallback configuration
Command History
Examples
The following example sets the date format:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# date-format mm-dd-yyRelated Commands
call-manager-fallback
Enables Cisco Unified SRST feature support and enters call-manager-fallback configuration mode.
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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-fallbackRouter(config-cm-fallback)# default-destination 40802Related Commands
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
Voice moh-group configuration (config-voice-moh-group)
Command History
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) description this is a moh group for sales
Related Commands
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. See Table 2 for a list of supported device IDs.
Command Default
Device ID is 0.
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
12.4(15)XZ
Cisco Unified CME 4.3
Cisco Unified SRST 4.3This 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
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
Related Commands
device-name
Assigns a name to a phone type in an ephone-type template.
load
Associates a type of phone with a phone firmware file.
type
Assigns the phone type to a SCCP 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
Command Default
No name is assigned to this phone type.
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
12.4(15)XZ
Cisco Unified CME 4.3
Cisco Unified SRST 4.3This 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
Related Commands
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 3 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
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.
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
Related Commands
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
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
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 potsRouter(config-dial-peer)# destination-pattern 1001Router(config-dial-peer)# voice-port 50/0/2and 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 potsRouter(config-dial-peer)# destination-pattern 4085551001Router(config-dial-peer)# voice-port 50/0/2Both 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-fallbackRouter(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-fallbackRouter(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-fallbackRouter(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-fallbackRouter(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..Related Commands
call-manager-fallback
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
show dial-peer voice
Displays information for voice dial peers.
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 configurationCommand History
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 4Router(config-register-pool)# digit collect kpmlRelated Commands
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
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
12.3(4)T
Cisco SIP SRST 3.0
This command was introduced.
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This command was added to Cisco Unified CME.
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
•
Examples
Cisco Unified CME
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 4Router(config-register-pool)# dtmf-relay rtp-nte sip-notifyCisco Unified SIP SRST
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 1application SIP.appincoming called-number 308voice-class codec 1dtmf-relay rtp-nteRelated Commands
dtmf-relay (voice over IP)
Specifies how an H.323 or SIP gateway relays DTMF tones between telephony interfaces and an IP network.
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
Command Default
No replacement number is created.
Command Modes
Voice emergency response location configuration (cfg-emrgncy-resp-location)
Command History
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 1elin 1 4085550100subnet 10.0.0.0 255.0.0.0subnet 2 192.168.0.0 255.255.0.0Related Commands
voice emergency response location
Creates a tag for identifying an ERL for Enhanced 911 Services.
subnet
Defines which IP phones are part of this ERL.
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
Command Default
No default ELIN number is created.
Command Modes
Voice emergency response settings configuration (cfg-emrgncy-resp-settings)
Command History
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 settingscallback 7500elin 4085550101expiry 120Related Commands
emergency response callback
To define a dial peer that is used for 911 callbacks from the PSAP, use the emergency response callback command in voice dial-peer configuration mode. To remove the definition of the dial peer as an incoming link from the PSAP, use the no form of this command.
emergency response callback
no emergency response callback
Syntax Description
This command has no arguments or keywords.
Command Default
The dial peer is not defined as an incoming link from the PSAP.
Command Modes
Dial-peer configuration (config-dial-peer)
Command History
Usage Guidelines
This command defines which dial peer is used for 911 callbacks from the PSAP. You can define multiple dial peers with this command.
Examples
The following example shows a dial peer defined as an incoming link from the PSAP. If 408 555-0100 is configured as the ELIN for an ERL, this dial peer recognizes that an incoming call from 408 555-0100 is a 911 callback.
dial-peer voice 100 potsincoming called-number 4085550100port 1/1:Ddirect-inward-dialemergency response callbackRelated Commands
emergency response location
To associate an emergency response location (ERL) for Enhanced 911 Services with a dial peer, ephone, ephone-template, voice register pool, or voice register template, use the emergency response location command in dial peer, ephone, ephone-template, voice register pool, or voice register template configuration mode. To remove the association, use the no form of this command.
emergency response location tag
no emergency response location tag
Syntax Description
tag
Unique number that identifies an existing ERL tag defined by the voice emergency response location command.
Command Default
No ERL tag is associated with a dial peer, ephone, ephone-template, voice register pool, or voice register template.
Command Modes
Dial-peer configuration (config-dial-peer)
Ephone configuration (config-ephone)
Ephone-template configuration (config-ephone-template)
Voice register pool configuration (config-register-pool)
Voice register template configuration (config-register-template)Command History
Usage Guidelines
Use this command to assign an ERL to phones individually. Depending on the type of phones (endpoints) that you have, you can assign an ERL to a phone's:
•
•
•
•
•
These methods of associating a phone with an ERL are alternatives to assigning a group of phones that are on the same subnet as an ERL.
The tag used by this command is an integer from 1 to 2147483647 and refers to an existing ERL tag that is defined by the voice emergency response location command. If the tag does not refer to a valid ERL configuration, the phone is not associated to an ERL. For IP phones, the IP address is used to find the inclusive ERL subnet. For phones is on a VoIP trunk or FXS/FXO trunk, the PSAP gets a reorder tone.
Examples
The following example shows how to assign an ERL to a phone's dial peer:
dial-peer voice 12 potsemergency response location 18The following example shows how to assign an ERL to a phone's ephone:
ephone 41emergency response location 22The following example shows how to assign an ERL to one or more SCCP phones:
ephone-template 6emergency response location 8The following example shows how to assign an ERL to a phone's voice register pool:
voice register pool 4emergency response location 21The following example shows how to assign an ERL to one or more SIP phones:
voice register template 4emergency response location 8Related Commands
emergency response zone
To define a dial peer that is used by the system to route emergency calls to a PSAP, use the emergency response zone command in voice dial-peer configuration mode. To remove the definition of the dial peer as an outgoing link to the PSAP, use the no form of this command.
emergency response zone zone-tag
no emergency response zone
Syntax Description
Command Default
The dial peer is not defined as an outgoing link to the PSAP. Therefore, E911 services are not enabled.
Command Modes
Dial-peer configuration (config-dial-peer)
Command History
Usage Guidelines
Use this command to specify that any calls using this dial peer are processed by the E911 software. To enable any E911 processing, the emergency response zone command must be enabled under a dial peer.
If no zone tag is specified, the system looks for a matching ELIN to the E911 caller's phone by searching each emergency response location that was configured using the emergency response location command.
If a zone tag is specified, the system looks for a matching ELIN using sequential steps according to the contents of the configured zone. For example, if the E911 caller's phone has an explicit ERL assigned, the system first looks for that ERL in the zone. If not found, it then searches each location within the zone according to assigned priority numbers, and so on. If all steps fail to find a matching ELIN, the default ELIN is assigned to the E911 caller's phone. If no default ELIN is configured, the E911 caller's automatic number identification (ANI) number is communicated to the Public Safety Answering Point (PSAP).
This command can be defined in multiple dial peers. The zone tag option allows only ERLs defined in that zone to be routed on this dial peer. Also, this command allows callers dialing the same emergency number to be routed to different voice interfaces based on the zone that includes their ERL.
Examples
The following example shows a dial peer defined as an outgoing link to the PSAP. Emergency response zone 10 is created and only calls from this zone are routed through 1/0/0.
dial-peer voice 911 potsdestination-pattern 9911prefix 911emergency response zone 10port 1/0/0Related Commands
ephone-type
To add a Cisco Unified IP phone type by defining an ephone-type template, use the ephone-type command in global configuration mode. To remove an ephone type, use the no form of this command.
ephone-type phone-type [addon]
no ephone-type phone-type
Syntax Description
Command Default
Ephone type is not defined.
Command Modes
Global configuration (config)
Command History
Usage Guidelines
This command adds a user-defined template for a phone type to a Cisco Unified CME system. This configuration template defines a set of attributes that describe the features of the new phone type. Use this command to add phone types that are not already defined in the system.
If you use a phone-type argument that matches a system-defined phone type, a message displays notifying you that the ephone-type is built-in and cannot be changed. For a list of system-defined phone types, see the type command.
Use the create cnf-files command for the new phone type to take effect.
Examples
The following example shows the Nokia E61 added with an ephone-type template, which is then assigned to ephone 2:
ephone-type E61
device-id 376
device-name E61 Mobile Phone
num-buttons 1
max-presentation 1
no utf8
no phoneload
!
!
telephony-service
load E61 SCCP61.8-2-2SR2S
max-ephones 100
max-dn 240
ip source-address 15.7.0.1 port 2000
cnf-file location flash:
cnf-file perphone
voicemail 8900
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Sep 25 2007 21:25:47
!
!
ephone 2
mac-address 001C.821C.ED23
type E61
button 1:2
Related Commands
expiry
To set the time after which emergency callback history expires, use the expiry command in voice emergency response settings configuration mode. To remove the number, use the no form of this command.
expiry time
no expiry
Syntax Description
time
Identifier (2-2880) in minutes for the maximum time the 911 caller history is available for callback.
Command Default
The default expiry time is 180 minutes.
Command Modes
Voice emergency response settings configuration (cfg-emrgncy-resp-settings)
Command History
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.
Usage Guidelines
Use this command to specify the amount of time (in minutes) to keep emergency caller history for each ELIN. The time can be an integer in the range of 2 to 2880 minutes. The default value is 180 minutes.
Examples
In this example, the ELIN (4085550101) defined in the voice emergency response settings configuration will be 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 settingscallback 7500elin 4085550101expiry 120Related Commands
extension-range
To define a range of extension numbers for a specific MOH group in Cisco Unified CME or Cisco Unified SRST, use the extension-range command in voice-moh-group configuration mode. To define a range of extension numbers for a specific directory number in Cisco Unified CME, use the extension-range command in ephone-dn configuration mode. To disable the extension-range command, use the no form of this command.
extension-range starting-extension to ending-extension
no extension-range starting-extension to ending-extension
Syntax Description
Command Default
No extension- range is configured.
Command Modes
Voice MOH group configuration (config-voice-moh-group)
Ephone-dn configuration (config-ephone-dn)Command History
Usage Guidelines
This command configured in voice moh-group configuration mode identifies MOH clients calling extension numbers specified under the extension range configured for a MOH group in Cisco Unified CME or Cisco Unified SRST. This command in ephone-dn configuration mode identifies MOH clients calling extension numbers specified under the extension range configured for a directory number in Cisco Unified CME
You can define multiple extension-ranges in the same MOH group or directory number.
The extension can be expressed in hexadecimal digits ranging from 0-9 or A-F and should not exceed the limit of 32 digits.
The starting-extension and ending-extension numbers must contain the same number of digits.
The ending extension number must be of a greater value than the starting extension number.
Extension-range for a MOH group must not overlap with any other extension-range configured in any other MOH group.
Note
Examples
The following example shows two extension ranges configured under voice moh-group 1:
Router(config)# voice moh-group 1
Router(config-voice-moh-group)# moh flahs:/minuet.wav
Router(config-voice-moh-group)# description Marketing
Router(cconfig-voice-moh-group)# extension range 1000 to 1999
Router(config-voice-moh-group)# extension range 3000 to 3999
Router(config-voice-moh-group)#
Related Commands
moh
Enables music on hold from an audio file.
voice-moh-group
Enters voice moh-group configuration mode.
external-ring (voice register global)
To specify the type of ring sound used on Cisco Session Initiation Protocol (SIP) or Cisco SCCP IP phones for external calls, use the external-ring command in voice register global configuration mode. To return to the default ring sound, use the no form of this command.
external-ring {bellcore-dr1 | bellcore-dr2 | bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
no external-ring
Syntax Description
Command Default
The default ring sound is an internal ring pattern.
Command Modes
Voice register global configuration
Command History
12.4(4)T
Cisco CME 3.4
Cisco SIP SRST 3.4This command was introduced.
Usage Guidelines
When set, this command defines varying ring tones so that you can discriminate between internal and external calls from Cisco SIP or Cisco SCCP IP phones.
Examples
The following example shows how to specify that Bellcore DR1 be used for external ringing on Cisco SIP IP phones:
Router(config)# voice register globalRouter(config-register-global)# external-ring bellcore-dr1Related Commands
h450 h450-2 timeout (voice service voip)
To specify timeout values for call transfers using the ITU-T H.450.2 standard, use the h450 h450-2 timeout command in H.323 voice service configuration mode. To return to the default values, use the no form of this command.
h450 h450-2 timeout {T1 | T2 | T3 | T4} milliseconds
no h450 h450-2 timeout {T1 | T2 | T3 | T4}
Syntax Description
T1
Timer for identification.
T2
Timer for call setup.
T3
Timer for response initiation.
T4
Timer for response setup.
milliseconds
Number of milliseconds. Range is from 500 to 60000.
Command Default
T1 timer is 2000 milliseconds.
T2 timer is 5000 milliseconds.
T3 timer is 5000 milliseconds.
T4 timer is 5000 milliseconds.Command Modes
H.323 voice service configuration
Command History
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
Use this command with Cisco SRST V3.0 or a later version.
This command is used primarily when the default settings for these timers do not match your network delay parameters. See the ITU-T H.450.2 specification for more information on these timers.
Examples
The following example defines a T1 timeout of 3000 milliseconds:
Router(config)# voice service voipRouter(conf-voi-serv)# h323Router(conf-serv-h323)# h450 h450-2 timeout T1 3000Related Commands
h323
Enables H.323 voice service configuration commands.
voice service
Enters voice-service configuration mode.
huntstop (call-manager-fallback)
To set the huntstop attribute for the dial peers associated with a Cisco Unified IP phone during Cisco Unified Communications Manager fallback, use the huntstop command in call-manager-fallback configuration mode. To disable huntstop, use the no form of this command.
huntstop [channel] 1-8
no huntstop
Syntax Description
Command Default
Huntstop is enabled by default. For octo-line mode, the default number is 8.
Command Modes
Call-manager-fallback configuration (config-cm-fallback)
Command History
Usage Guidelines
In call-manager-fallback configuration mode, the huntstop attribute by default is set uniformly to all Cisco Unified IP phone lines (for example, to all or to none).
Note
Examples
The following example shows how to disable huntstop for all Cisco Unified IP phones:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# no huntstopThe following example shows that octo-line is enabled, the maximum directory numbers is 12, and huntstop is limited at 6 channels:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# max-dn 12 octo-line 6
Related Commands
id (voice register pool)
To explicitly identify a locally available individual Cisco SIP IP phone, or when running Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST), set of Cisco SIP IP phones, use the id command in voice register pool configuration mode. To remove local identification, use the no form of this command.
id {network address mask mask | ip address mask mask | mac address}
no id {network address mask mask | ip address mask mask | mac address}
Syntax Description
Command Default
None
Command Modes
Voice register pool configuration
Command History
Usage Guidelines
Configure the id (voice register pool) command before any other voice register pool commands.
The id command allows explicit identification of an individual Cisco SIP IP phone to support a degree of authentication, which is required to accept registrations, based upon the following:
•
•
When the mac address keyword and argument are used, the IP phone must be in the same subnet as that of the router's LAN interface, such that the phone's MAC address is visible in the router's ARP cache. Once a MAC address is configured for a specific voice register pool, remove the existing MAC address before changing to a new MAC address.
Note
Examples
The following is partial sample output from the show running-config command. The id command identifies the MAC address of a particular Cisco IP phone. The output shows that voice register pool 1 has been set up to accept SIP Register messages from a specific IP phone through the use of the id command.
voice register pool 1id mac 0030.94C2.A22Apreference 5cor incoming call91 1 91011translate-outgoing called 1proxy 10.2.161.187 preference 1 monitor probe icmp-pingalias 1 94... to 91011 preference 8voice-class codec 1Related Commands
incoming called-number (voice register pool)
To apply incoming called-number parameters to dynamically created dial peers, use the incoming called-number command in voice register pool configuration mode. To remove incoming called-number parameters from a dial peer, use the no form of this command.
incoming called-number [number]
no incoming called-number
Syntax Description
Command Default
None
Command Modes
Voice register pool configuration
Command History
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.
Usage Guidelines
The id (voice register pool) command must be configured before any other voice register pool commands, including the incoming called-number command. The id command identifies a locally available individual Cisco SIP IP phone or a set of Cisco SIP IP phones.
Examples
The following is partial sample output from the show running-config command that applies the prefix 308 to dynamically created dial peers:
voice register pool 1application SIP.appincoming called-number 308voice-class codec 1Related Commands
ip qos dscp (call-manager-fallback)
To set the Differentiated Services Code Point (DSCP) for marking the quality of service (QoS) requirements for each packet, use the ip qos dscp command in call-manager-fallback configuration mode. To reset to the default value, use the no form of this command.
ip qos dscp {{number | af | cs | default | ef} {media | service | signaling | video}}
no ip qos dscp {{number | af | cs | default | ef} {media | service | signaling | video}}
Syntax Description
Defaults
DSCP for media is ef.
Command Modes
Call-manager-fallback configuration (config-cm-fallback)
Command History
Usage Guidelines
This command allows you to set different priority levels for different types of network traffic sent by the Cisco Unified SRST router. Differentiated Services is a method of prioritizing specific network traffic based on the QoS specified by each packet.
Because Cisco Unified IP Phones get their DSCP information from the configuration file that is downloaded to the device, and Cisco Unified SRST does not download configuration files to phones during fallback, only the value set with the media keyword takes effect for SCCP phones.
If the DSCP is configured for the gateway interface using the service-policy command or in the dial peer using the ip qos dscp command, the value set with those commands takes precedence over the DSCP value configured with this command.
Examples
The following example shows DSCP set for different types of packets.
call-manager-fallback
ip qos dscp cs2 media
ip qos dscp cs1 signal
ip qos dscp cs3 video
ip qos dscp cs4 service
Related Commands
ip qos dscp
Sets the DSCP for QoS in a dial peer.
service-policy
Assigns a policy map to an interface that will be used as the service policy for the interface.
ip source-address (call-manager-fallback)
To enable the SRST router to receive messages from Cisco IP phones through the specified IP addresses and ports, use the ip source-address command in call-manager-fallback configuration mode. To disable the router from receiving messages from Cisco IP phones, use the no form of this command.
ip source-address ip-address [port port] [any-match | strict-match]
no ip source-address [ip-address port port] [any-match | strict-match]
Syntax Description
Command Default
Default port number: 2000
Default server address match: any-matchCommand Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The ip source-address command is a mandatory command. The fallback subsystem does not start if the IP address of the Ethernet port to which the phones are connected (typically the ethernet interface of the local SRST gateway) is not provided. If the port number is not provided, the default value (2000) is used.
Use the any-match keyword to instruct the router to permit Cisco IP phone registration even when the IP server address used by the phone does not match the IP source address. This option can be used to allow registration of Cisco IP phones on different subnets or those with different default DHCP routers or different TFTP server addresses.
Use the strict-match keyword to instruct the router to reject Cisco IP phone registration attempts if the IP server address used by the phone does not exactly match the source address. By dividing the Cisco IP phones into groups on different subnets and giving each group different DHCP default-router or TFTP server addresses, this option can be used to restrict the number of Cisco IP phones allowed to register.
The ip source-address command enables a router to receive messages from Cisco IP phones through the specified IP addresses and port. If the router receives a registration request from a Cisco IP phone, the router in return requests the phone configuration and dial-plan information from the Cisco IP phone. This data is stored locally in the memory of the router and is used to create voice-port and dial-plan information. The voice-port and dial-plan information is used to handle telephony calls to and from the Cisco IP phone if the Cisco Unified Communications Manager is unreachable.
Examples
The following example shows how to set the IP source address and port:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# ip source-address 10.6.21.4 port 2002 strict-matchRelated Commands
call-manager-fallback
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
ip source-address (credentials)
To enable the Cisco Unified CME or SRST router to receive credential service messages through the specified IP address and port, use the ip source-address command in credentials configuration mode. To disable the router from receiving messages, use the no form of this command.
ip source-address ip-address [port [port]]
no ip source-address
Syntax Description
Command Default
Cisco Unified CME default port number: 2444
Cisco Unified SRST default port number: 2445Command Modes
Credentials configuration
Command History
Usage Guidelines
Cisco Unified CME
This command is used with Cisco Unified CME phone authentication to identify a Cisco Unified CME router on which a CTL provider is being configured.
Cisco Unified SRST
The ip source-address command is a mandatory command to enable secure SRST. If the port number is not provided, the default value (2445) is used. The IP address is usually the IP address of the secure SRST router.
Examples
Cisco Unified CME
The following example creates a CTL provider on a Cisco Unified CME router that is not running the CTL client.
Router(config)# credentials
Router(config-credentials)# ip source-address 172.19.245.1 port 2444
Router(config-credentials)# trustpoint ctlpv
Router(config-credentials)# ctl-service admin user4 secret 0 c89L8o
Cisco Unified SRST
The following example enters credentials configuration mode and sets the IP source address and port:
Router(config)# credentialsRouter(config-credentials)# ip source-address 10.6.21.4 port 2445Related Commands
keepalive (call-manager-fallback)
To configure the time interval between successive keepalive messages from Cisco IP phones, use the keepalive command in call-manager-fallback configuration mode. To restore to the default interval, use the no form of this command.
keepalive seconds
no keepalive seconds
Syntax Description
seconds
Keepalive message transmission interval, in seconds. The valid range is from 10 to 65535. The default timeout value is 30.
Command Default
30 seconds
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The keepalive command configures the time interval between successive keepalive messages from Cisco IP phones to the Cisco Unified SRST router. If the router fails to receive three successive keepalive messages, it considers the Cisco IP phone to be out of service until the phone reregisters.
Note
Examples
The following example sets the keepalive timeout value to 60 seconds:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# keepalive 60Related Commands
call-manager-fallback
Enables Cisco Unified SRST feature support and enters call-manager-fallback configuration mode.
limit-dn (call-manager-fallback)
To specify the maximum number of lines available on each Cisco IP phone, use the limit-dn command in call-manager-fallback configuration mode. To return to the default setting, use the no form of this command.
limit-dn phone-type max-lines
no limit-dn phone-type
Syntax Description
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The limit-dn command specifies the maximum number of lines (directory numbers) available for each Cisco IP phone type.
Note
The range of the maximum number of lines is from 1 to 34. If there is any active phone with a number greater than the specified limit, warning information is displayed for phone reset.
Examples
The following example shows how to set a directory number limit of 2 for the Cisco Unified IP Phone 7910:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# limit-dn 7910 2Related Commands
call-manager-fallback
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
location (voice emergency response zone)
To include a location within an emergency response zone, use the location command in voice emergency response zone mode. To assign specific priorites to the locations, use the priority tag. To remove the location, use the no form of this command.
location location-tag [priority <1-100>]
no location location-tag
Syntax Description
location-tag
Identifier for the emergency response zone location.
priority 1-100
Identifier (1-100) for the priority ranking of locations, 1 being the highest priority.
Command Modes
Voice emergency response zone configuration (cfg-emrgncy-resp-zone)
Command History
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.
Usage Guidelines
This command creates locations within emergency response zones. The tag must be the same as the tag that is defined using the voice emergency response location command. This allows routing of 911 calls to different PSAPs. Priority is optional and allows searching locations in a specified priority order. If there are locations with assigned priorities and locations configured without priorities, the prioritized locations are searched before those without an assigned priority.
Examples
The following example shows an assignment of ERLs to two zones, 10 and 11, to route callers to two different PSAPs. The locations for ERLs in zone 10 are searched in sequential order for a phone address match. The calls from zone 10 have an ELIN from ERLs 8, 9, and 10. The calls from zone 11 have an ELIN from ERLs 2, 3, 4, and 5. The locations for ERLs in zone 11 have priorities assigned and is searched in order of the assigned priority and not the ERL tag number.
voice emergency response zone 10location 8location 9location 10voice emergency response zone 11location 5 priority 1location 3 priority 2location 4 priority 3location 2 priority 10Related Commands
logging (voice emergency response settings)
To enable sylog messages to capture emergency call data, use the logging command in voice emergency response settings configuration mode. To disable logging, use the no form of this command.
logging
no logging
Syntax Description
This command has no arguments or keywords.
Command Default
This command is enabled by default.
Command Modes
Voice emergency response settings configuration (cfg-emrgncy-resp-settings)
Command History
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.
Usage Guidelines
This command enables syslog messages to be announced for every 911 emergency call that is made. The syslog messages can be used by third party applications to send pager or e-mail notifications to an in-house support number. This command is optional and is enabled by default.
Examples
In this example, the ELIN (4085550101) defined in the voice emergency response settings configuration is used if the 911 caller's IP phones 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. The outbound 911 calls do not emit a syslog message to the logging facility (for example, a local buffer, console, or remote host).
voice emergency response settingscallback 7500elin 4085550101expiry 120no loggingRelated Commands
max-conferences (call-manager-fallback)
To set the maximum number of simultaneous three-party conferences supported by the router, use the max-conferences command in call-manager-fallback configuration mode. To return to the default number of conferences, use the no form of this command.
max-conferences max-no-of-conferences [gain -6 | 0 | 3 | 6]
no max-conferences max-no-of-conferences
Syntax Description
Command Default
The default is half of the number of maximum simultaneous three-party conferences allowed per platform.
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The max-conferences command supports three-party conferences for local and on-net calls only when all conference participants are using the G.711 codec. Conversion between G.711 u-law and a-law is supported. Mixing of the media streams is supported by the Cisco IOS processor. The maximum number of simultaneous conferences is limited to the platform-specific maximum.
The gain keyword's functionality is applied to inbound audio packets, so conference participants can more clearly hear a remote PSTN or VoIP caller joining their call. Note that this functionality cannot discriminate between a remote VoIP/foreign exchange office (FXO) source, which requires a volume gain, and a remote VoIP/IP phone, which does not require a volume gain and may therefore incur some sound distortions.
Examples
The following example sets the maximum number of conferences for a Cisco IP phone to 4 and configures a gain of 6 db for inbound audio packets from remote PSTN or VoIP calls joining a conference:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# max-conferences 4 gain 6Related Commands
call-manager-fallback
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
max-dn (call-manager-fallback)
To set the maximum possible number of directories or virtual voice ports that can be supported by a router and to activate dual-line mode, octo-line mode, or both modes, use the max-dn command in call-manager-fallback configuration mode. To return to the default number of directories or virtual voice ports and to deactivate the dual-line mode or octo-line mode, use the no form of this command.
max-dn max-no-of-directories [dual-line | octo-line] [preference preference-order]
[number octo-line]no max-dn
Syntax Description
Command Default
0 directory numbers, 1 channel per virtual port, 8 channels per virtual port, 0 preference, 8 numbers
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The max-dn command limits the number of Cisco Unified IP phone directory numbers or virtual voice ports available on the router.
The dual-line keyword facilitates call waiting, call transfer, and conference functions by allowing two calls to occur on one line simultaneously. In dual-line mode, all Cisco Unified IP phones on the Cisco Unified SRST router support two channels per virtual voice port.
The octo-line keyword facilitates call waiting, call transfer, and conference functions by allowing eight calls to occur on one line simultaneously. In octo-line mode, all Cisco Unified IP phones on the Cisco Unified SRST router support eight channels per virtual voice port.
Note
During Cisco Unified SRST registration, a dial peer is created and that dial peer includes a default preference. The preference keyword allows you to change the default value, if desired.
Setting the preference enables the desired dial peer to be selected when multiple dial peers within a hunt group are matched for a dial string.
The alias command also has a preference keyword that sets alias command preference values. Setting the alias command preference keyword allows the default preference set with the max-dn command to be overridden. When configuring call rerouting with the alias command, set the preference keyword of the max-dn command to a higher numeric preference value than the preference set with the alias command.
Examples
The following example sets the maximum number of directory numbers or virtual voice ports to 12, activates dual-line mode, activates octo-line mode, and sets the maximum number of dns for octo-mode to 6:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# max-dn 12 dual-line preference 1 octo-line 6Related Commands
max-ephones (call-manager-fallback)
To configure the maximum number of Cisco IP phones that can be supported by a router, use the max-ephones command in call-manager-fallback configuration mode. To return to the default number of Cisco IP phones, use the no form of this command.
max-ephones max-no-of-phones
no max-ephones
Syntax Description
max-no-of-phones
Maximum number of Cisco IP phones supported by the router. The maximum possible number is platform-dependent; refer to Cisco IOS command-line interface (CLI) help. The default is 0.
Command Default
0 Cisco IP phones
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The max-ephones command limits the number of Cisco IP phones supported on the router.
Note
Examples
The following example sets the maximum number of Cisco IP phones for a Cisco router to 24:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# max-ephones 24Related Commands
call-manager-fallback
Enables Cisco Unified SRST feature support and enters call-manager-fallback configuration mode.
maximum bit-rate (cm-fallback-video)
To set the maximum IP phone video bandwidth, use the maximum bit-rate command in call-manager-fallback video configuration mode. To restore the default maximum bit-rate, use the no form of this command.
maximum bit-rate value
no maximum bit-rate
Syntax Description
value
Sets the maximum IP phone video bandwidth, in kbps. The range is 0 to 10000000. The default value is 10000000.
Command Default
Maximum bit-rate is 1000000.
Command Modes
Call-manager-fallback video configuration
Command History
12.4(4)XC
Cisco Unified SRST 4.0
This command was introduced.
Usage Guidelines
Use this command to set the maximum bit-rate for all video-capable phones in a Cisco Unified SRST system.
Examples
The following example sets a maximum bit-rate of 256 kbps.
Router(config)#call-manager-fallbackRouter(config-call-manager-fallback)# video
Router(conf-cm-fallback-video)# maximum bit-rate 256Related Commands
maximum bit-rate
(telephony-service)Sets the maximum bit-rate for all video-capable phones associated with a Cisco Unified CME router.
max-dn (voice register global)
To set the maximum number of SIP phone directory numbers (extensions) that are supported by a Cisco router, use the max-dn command in voice register global configuration mode. To reset to the default, use the no form of this command.
max-dn max-directory-numbers
no max-dn
Syntax Description
Command Default
Before Cisco Unified CME 7.1 and Cisco Unified SIP SRST 7.1, default is maximum number supported by platform.
In Cisco Unified CME 7.1 and Cisco Unified SIP SRST 7.1 and later versions, default is 0.
Command Modes
Voice register global configuration (config-register-global)
Command History
Usage Guidelines
This command limits the number of SIP phone directory numbers (extensions) available in a Cisco Unified CME system. The max-dn command is platform specific. It defines the limit for the voice register dn command. The max-pool command similarly limits the number of SIP phones in a Cisco CME system.
You can increase the number of allowable extensions to the maximum; but after the maximum allowable number is configured, you cannot reduce the limit without rebooting the router. You cannot reduce the number of allowable extensions without removing the already-configured directory numbers with dn-tags that have a higher number than the maximum number to be configured.
Note
Examples
The following example shows how to set the maximum number of directory numbers to 48:
Router(config)# voice register global
Router(config-register-global)# max-dn 48
Related Commands
max-pool (voice register global)
To set the maximum number of Session Initiation Protocol (SIP) voice register pools that are supported in Cisco Unified SIP SRST or Cisco Unified CME, use the max-pool command in voice register global configuration mode. To reset the maximum number to the default, use the no form of this command.
max-pool max-voice-register-pools
no max-pool
Syntax Description
Command Default
Before Cisco Unified CME 7.1 and Cisco Unified SIP SRST 7.1, default is maximum number supported by platform.
In Cisco Unified CME 7.1 and Cisco Unified SIP SRST 7.1 and later versions, default is 0.Command Modes
Voice register global configuration (config-register-global)
Command History
Usage Guidelines
This command limits the number of SIP phones supported by Cisco Unified CME. The max-pool command is platform specific and defines the limit for the voice register pool command.
The max-dn command similarly limits the number of directory numbers (extensions) in Cisco Unified CME.
You can increase the number of phones; but after the maximum allowable number is configured, you cannot reduce the limit of the SIP phones without rebooting the router.
Note
Examples
The following example shows how to set the maximum number of Cisco SIP IP phones in Cisco Unified SIP SRST or Cisco Unified CME to 24:
Router(config)# voice register global
Router(config-register-global)# max-pool 24
Related Commands
max-dn (voice register global)
Set the maximum number of SIP phone directory numbers (extensions) that are supported by a Cisco Unified CME router.
max-presentation
To set the number of call presentation lines supported by a phone type, use the max-presentation command in ephone-type configuration mode. To reset to the default, use the no form of this command.
max-presentation number
no max-presentation
Syntax Description
number
Number of presentation lines. Range: 1 to 100. Default: 0. See Table 4 for the number of presentation lines supported by each phone type.
Command Default
No display lines are supported by the phone type.
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
Usage Guidelines
This command defines the number of presentation lines that are supported for the type of phone being added with an ephone-type template.
Table 4 Supported Values for Ephone-Type Commands
Cisco Unified IP Conference Station 7937G
431
1
6
Nokia E61
376
1
1
Examples
The following example shows that 1 presentation line is specified for the Nokia E61 when creating the ephone-type template.
Router(config)# ephone-type E61
Router(config-ephone-type)# max-presentation 1
Related Commands
max registrations (voice register pool)
Note
To set the maximum number of registrations accepted by the voice register pool, use the max registrations command in voice register pool configuration mode. To disable registration setup, use the no form of this command.
max registrations value
no max registrations
Syntax Description
value
Digit, beginning with 0, that represents the maximum number of registrations. The maximum registration value is platform dependent.
Defaults
The maximum number of IP phones that can be configured per platform
Command Modes
Voice register pool configuration
Command History
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(4)T
This command was removed.
Usage Guidelines
The id (voice register pool) command must be configured before any other voice register pool commands, including the max registrations command. The id command identifies a locally available individual Cisco SIP IP phone or sets of Cisco SIP IP phones.
If two phones attempt to register the same phone number, only the first phone can register the number. You can control which phone is accepted by using multiple voice register pools. In general, the best usage is one pool per phone; with multiple pools, some flexibility is granted.
Examples
The following partial sample output from the show running-config command shows that 5 is the maximum number of SIP telephone registrations accepted.
voice register pool 3id network 10.2.161.0 mask 255.255.255.0number 1 95... preference 1cor outgoing call95 1 95011max registrations 5voice-class codec 1Related Commands
moh-file-buffer (cm-fallback)
To specify a MOH file buffer size, use the moh-file-buffer command in call-manager-fallback configuration mode. To delete the moh-file-buffer size, use the no form of this command.
moh-file-buffer file_size
no moh-file-buffer
Syntax Description
Command Default
No moh-file buffer is configured.
Command Modes
Call-manager-fallback configuration (config-cm-fallback)
Command History
15.0(1)XA
Cisco Unified SRST 8.0
This command was introduced.
15.1(1)T
Cisco Unified SRST 8.0
This command was integrated into Cisco IOS Release 15.1(1)T.
Usage Guidelines
This command allows to set a buffer MOH file size limit for new MOH files. You can allocate a MOH file buffer size between 64 KB (8 seconds ) and 10000 KB (20 minutes, approximately). A large buffer size is desirable to cache the largest MOH file and better MOH performance. During memory allocation the buffer size is aligned to 16KB. For more information on MOH file caching, see show ephone moh command.
The default maximum file buffer size is 64 KB. If the MOH file size is too large, it cannot be cached and the buffer size falls back to 64 KB.
Note
Examples
The following example shows a moh-file-buffer size of 2000 KB assigned for future moh files under the call-manager-fallback configuration mode.
!!!!call-manager-fallbackmax-conferences 8 gain -6transfer-system full-consultmoh-file-buffer 2000!!line con 0exec-timeout 0 0line aux 0--More--Related Commands
moh (call-manager-fallback)
To enable music on hold (MOH), use the moh command in call-manager-fallback configuration mode. To disable music on hold, use the no form of this command.
moh filename
no moh filename
Syntax Description
Command Default
MOH is enabled.
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The moh command allows you to specify the .au and .wav format music files that are played to callers who have been put on hold. MOH works only for G.711 calls and on-net VoIP and PSTN calls. For all other calls, callers hear a periodic tone. For example, internal calls between Cisco IP phones do not get MOH; instead callers hear a tone.
Note
MOH can be used as a fallback MOH source when using MOH live feed. See the moh-live (call-manager-fallback) command for more information.
Examples
The following example enables MOH and specifies the music files:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# moh minuet.wavRouter(config-cm-fallback)# moh minuet.auRelated Commands
moh (voice moh-group)
To enable music on hold (MOH) for a MOH group, use the moh command in voice moh-group configuration mode. To disable music on hold, use the no form of this command.
moh filename
no moh filename
Syntax Description
Command Default
No MOH is enabled
Command Modes
Voice moh-group configuration (config-voice-moh-group)
Command History
Usage Guidelines
The moh command allows you to specify the .au and .wav format music files that are played to callers who have been put on hold. MOH works only for G.711 calls and on-net VoIP and PSTN calls. For all other calls, callers hear a periodic tone. You must provide the directory and filename of the MOH file in URL format. For example: moh flash:/minuet.au
Note
Examples
The following example enables MOH for voice moh group 1 and specifies the music files:
Router(config)#
Router(config)#voice moh-group 1
Router(config-voice-moh-group)# moh flash:/minuet.wav
Related Commands
moh-live (call-manager-fallback)
To specify that a particular telephone number is to be used for an outgoing call that is to be the source for a music on hold (MOH) stream for SRST, use the moh-live command in call-manager-fallback configuration mode. To disable the source for the MOH stream, use the no form of this command.
moh-live dn-number calling-number out-call outcall-number
no moh-live dn-number calling-number out-call outcall-number
Syntax Description
Command Default
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
The moh-live command in SRST mode provides live-feed MOH streams from an audio device connected to an E&M or FXO port to Cisco IP phones.
During SRST, do not change an existing moh-live command. Instead, first remove the existing command, and then add in the new command.
To configure MOH from a live feed, you first establish a voice port and dial peer for the call and then create a "dummy" phone or directory number. The dummy number allows for making and receiving calls, but the number is not assigned to a physical phone. It is that number that the MOH system automatically dials to establish the MOH feed.
The moh-live command allocates one of the virtual voice ports from the pool of virtual voice ports created by the max-dn command. The virtual voice port places an outgoing call to the dummy number; that is, the directory number specified in the moh-live command. The audio stream obtained from the MOH call provides the music-on-hold audio stream.
To activate MOH live feed, set up and connect the physical voice port. After setting up the voice port, create a dial peer and give the voice port a directory number with the destination-pattern command. The directory number is the number that the system uses to access the MOH. To establish the MOH feed, connect the music source, such as a CD player to autodial the directory number.
MOH can be used as a fallback MOH source when using MOH live feed. See the moh (call-manager-fallback) command for more information.
Examples
The following example configures MOH from a live feed. Note that the dial peer references the E&M port that was set with the voice-port command and that the dial-peer number (7777) matches the out-call value of the moh-live command.
...voice-port 1/0/0 input gain 3 auto-cut-through operation 4-wire signal immediate ! dial-peer voice 7777 pots destination-pattern 7777 port 2/0/0 ! ! call-manager-fallback max-conferences 8 max-dn 1 moh-live dn-number 3333 out-call 7777 ! ...Related Commands
multicast moh (call-manager-fallback)
To enable continuous IP multicast output of music on hold (MOH) from a flash MOH file in a branch Cisco Unified SRST router, use the multicast moh command in call-manager-fallback configuration mode. To disable continuous IP multicast output of MOH from a flash MOH file in a branch Cisco Unified SRST router, use the no form of this command.
multicast moh multicast-address port port [route ip-address-list]
no multicast moh multicast-address port port [route ip-address-list]
Syntax Description
Command Default
If the route keyword is not in the multicast moh command, the ip source-address command value configured for Cisco Unified SRST will be used.
Command Modes
Call-manager-fallback configuration
Command History
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
MOH multicast operates during Cisco Unified Communications Manager fallback and normal Cisco Unified Communications Manager service.
Because all branch IP phones are instructed by the central Cisco Unified Communications Manager to select and multicast a specific MOH feed, you must configure the IP address and port header values to match the MOH multicast IP address and port used by the central Cisco Unified Communications Manager's MOH. This makes the MOH from Cisco Unified SRST router appear as if it were coming from the central Cisco Unified Communications Manager.
MOH multicast operates with both MOH and MOH live input sources.
Examples
The following example provides multicast MOH output to the IP phones local to SRST. The MOH packets are broadcast to the 239.10.20.30 multicast address using port 2000. The MOH packets use the ip source-address command value configured for Cisco Unified SRST. MOH packets are output only through the router interface that matches the SRST router's IP source-address:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# moh music-on-hold-file.auRouter(config-cm-fallback)# multicast moh 239.10.20.30 port 2000The following example provides multicast MOH output to the IP phones local to SRST. The MOH packets are broadcast to the 239.10.20.30 multicast address using port 2000. The MOH packets use the ip source-address command value configured for Cisco Unified SRST. MoH packets are output only through the router interfaces that match the IP addresses listed using the route keyword option:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# moh music-on-hold-file.auRouter(config-cm-fallback)# multicast moh 239.10.20.30 port 2000 route 10.10.20.1 10.10.21.1 10.10.22.1 10.10.23.1Related Commands
mwi expires (call-manager-fallback)
Effective with Cisco IOS Releases 12.3(11)T7 and 12.4, the mwi expires command was replaced with the mwi-server command in SIP user-agent configuration mode.
To set the expiration timer for registration for the message-waiting indication (MWI) client or server, use the mwi expires command in call-manager-fallback configuration mode. To disable the timer, use the no form of this command.
mwi expires seconds
no mwi expires seconds
Syntax Description
Command Default
86400 seconds (24 hours)
Command Modes
Call-manager-fallback configuration
Command History
Examples
The following example sets the expiration timer to 1000 seconds:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# mwi expires 1000Related Commands
mwi relay (call-manager-fallback)
Enables a Cisco Unified SRST router to relay MWI notification to remote Cisco IP phones.
mwi reg-e164 (call-manager-fallback)
To register E.164 numbers rather than extension numbers with a Session Interface Protocol (SIP) proxy or registrar, use the mwi reg-e164 command in call-manager-fallback configuration mode. To return to the default, use the no form of this command.
mwi reg-e164
no mwi reg-e164
Syntax Description
This command has no keywords or arguments.
Command Default
Registering extension numbers with the SIP proxy or registrar.
Command Modes
Call-manager-fallback configuration
Command History
12.3(11)T7
12.4Cisco SRST 3.3
This command was introduced.
Usage Guidelines
This command is used when setting up extensions to use an external SIP-based message-waiting indication (MWI) server. The mwi-server command in SIP user-agent configuration mode specifies other settings for MWI service.
Examples
The following example specifies that E.164 numbers should be used for registration with the SIP proxy or registrar:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# mwi reg-e164Related Commands
mwi-server (SIP user-agent)
Specifies voice-mail server settings on a voice gateway or user agent (UA).
mwi relay (call-manager-fallback)
To enable a Cisco Unified SRST router to relay message-waiting indication (MWI) notification to remote Cisco IP phones, use the mwi relay command in call-manager-fallback configuration mode. To disable MWI relay, use the no form of this command.
mwi relay
no mwi relay
Syntax Description
This command has no arguments or keywords.
Command Default
MWI is not enabled.
Command Modes
Call-manager-fallback configuration
Command History
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
Use this command to enable the Cisco Unified SRST router to relay MWI notification to remote Cisco IP phones. The router at the central site acts as a notifier after this command is used.
Examples
The following example enables MWI relay:
Router(config)# call-manager-fallbackRouter(config-cm-fallback)# mwi relayRelated Commands
mwi expires (call-manager- fallback)
Sets the expiration timer for registration for the client or the server.
mwi sip-server (call-manager-fallback)
Effective with Cisco IOS Releases 12.3(11)T7 and 12.4, the mwi sip-server command was replaced with the mwi-server command in SIP user-agent configuration mode and the mwi reg-e164 command in call-manager-fallback configuration mode.
To configure the IP address and port number for an external SIP-based message-waiting indication (MWI) server, use the mwi sip-server command in call-manager-fallback configuration mode. To disable the MWI server functionality, use the no form of this command.
mwi sip-server ip-address [transport tcp | transport udp] [port port-number] [reg-e164] [unsolicited]
no mwi sip-server ip-address
Syntax Description
Command Default
Transport layer protocol: TCP
Port number: 5060 (SIP standard port)
Registration: with an extension numberCommand Modes
Call-manager-fallback configuration
Command History
Usage Guidelines
Use this command to configure the IP address of an external SIP MWI server.
The transport tcp keyword is the default setting. The transport udp keyword allows you to integrate with a SIP MWI client. The optional port keyword is used to specify a port number other than 5060, the default. The default registration is with an extension number, so the reg-e164 keyword allows you to register with an E.164 ten-digit number.
Examples
The following example sets MWI for the SIP server and sets individual ephone-dn extension numbers to the MWI SIP server's notification list:
Router(config) call-manager-fallbackRouter(config-cm-fallback) mwi sip-server 192.168.0.5 transport udpRelated Commands
mwi relay (call-manager-fallback)
Enables a Cisco Unified SRST router to relay MWI notification to remote Cisco IP phones.
