Usage Guidelines

Use this command to specify the MAC address of a specific Cisco IP phone in order to physically identify the Cisco IP phone in a Cisco CME configuration. The MAC address of each Cisco IP phone is printed on a sticker that is placed on the bottom of the phone.

If you choose to register phones before configuring them, the mac-address command can be used during configuration without entering the mac-address argument. The Cisco CME system detects MAC addresses and automatically populates phone configurations with their corresponding MAC addresses and phone types. This capability is not supported for voice-mail ports and is supported only by Cisco CME 3.0 and later versions. To use this capability, enable Cisco CME by using the following commands: max-ephones , max-dn , create cnf-files , and ip source-address. After these commands have been used, phones can start to register. Then, when you are configuring a registered ephone and you use the mac-address command with no argument, the MAC address of the phone is automatically read into the configuration. The equivalent functionality is available through the Cisco CME graphic user interface (GUI).

If you choose to configure phones before registering them, the MAC address for each ephone must be entered during configuration.

Usage Guidelines

This command defines the MAC address of the Cisco voice gateway that downloads its XML configuration file from Cisco Unified CME using the Autoconfiguration feature.

Usage Guidelines

When Cisco Unified CME diverts a call, it captures the reroute information which will be used to compose a reroute request. A dial-peer match will be performed against the diverted-to number. If this is the voice mail pilot number and the mailbox-selection command has been used to install a policy, the reroute information will be amended as directed by the command. The originator will pick up the modified reroute request, build the diversion information and include it in the new diverted call to the voice-mail pilot number.

This command should be used on the outbound dial peer for the pilot number of the voice-mail system.

This command might not work properly in certain network topologies, including the following cases:

• When the last redirecting endpoint is not hosted on Cisco Unified CME. This rarely occurs with a PBX.
• When a call is forwarded across several SIP trunks. Multiple SIP Diversion Headers (stacking hierarchy) are not supported in Cisco IOS software.
• When a call is forwarded across non Cisco voice gateways that do not support the optional H450.3 originalCalledNr field.

Usage Guidelines

This command sets the policy for selecting a mailbox for diverted calls.

This command is used on the ephone-dn associated with the voice-mail pilot number.

This command can only be used with SCCP phones.

This command might not work properly in certain network topologies, including the following cases:

• When the last redirecting endpoint is not hosted on Cisco Unified CME. This may rarely occur with a PBX.
• When a call is forwarded across several SIP trunks. Multiple SIP Diversion Headers (stacking hierarchy) are not supported in Cisco IOS software.
• When a call is forwarded across non Cisco voice gateways that do not support the optional H450.3 originalCalledNr field.

Usage Guidelines

This command limits the maximum number of calls, both incoming and outgoing, that can be active on each octo-line directory number on an SCCP phone. This command applies to all octo-line directory numbers on the phone.

This command must be set to a value that is more than or equal to the value set with the busy-trigger-per-button command.

For phones that do not support octo-line directory numbers such as the Cisco Unified IP Phone 7902, 7920, or 7931, and analog phones connected to the Cisco VG224 or Cisco ATA, we recommend that you set the max-calls-per-button command to 2. Otherwise, after the phone type is identified with either the type command or during phone registration, this command is automatically set to 2.

If you use an ephone template to apply a command to an ephone and you also use the same command in ephone configuration mode for the same ephone, the value that you set in ephone configuration mode has priority.

Usage Guidelines

This 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 mu-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 maximums.

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.

Usage Guidelines

The max-dn command limits the number of extensions (ephone-dns) available in a Cisco Unified CME system. The maximum number of ephone-dns that you can create depends on the software version, router platform, and amount of memory that you have installed. Type ? to display range.

The max-ephones command similarly limits the number of IP phones in a Cisco Unified CME system.

Note	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.

If registration with an H.323 gatekeeper or SIP proxy is enabled globally (the default), you can override the setting per extension by using the no-reg keyword in the number command for individual ephone-dns.

After using this command, you can provision individual extensions using the Cisco Unified CME graphic user interface (GUI) or the router CLI in ephone-dn configuration mode.

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	This command can also be used for Cisco Unified SIP SRST.

Usage Guidelines

The max-ephones command limits the number of Cisco IP phones supported on the router. The maximum number you can set is platform- and version-dependent. Use CLI help to determine the maximum number of ephones you can set, as shown in this example:

Router(config-telephony)# max-ephones ?
  <1-48>  Maximum phones to support

The max-dn command similarly limits the number of extensions (ephone-dns) in a Cisco CME system.

Note	You can increase the number of phones; but after the maximum allowable number is configured, you cannot reduce the limit of the Cisco IP phones without rebooting the router.

After using this command, configure phones by using the Cisco CME graphic user interface (GUI) or the router CLI in ephone configuration mode.

Usage Guidelines

This command creates an idle-duration timer for automatically logging a user out from Extension Mobility. The timer monitors the phone and if the specified maximum idle time is exceeded, the EM manager logs out the user. Typically this command is used to log out users who fail to manually log out of Extension Mobility before leaving a phone.

The call history record is automatically cleared when a user logs out from an Extension Mobility phone. To disable Automatic Clear Call History on all Extension Mobility phones, use the keep call-history command in telephony-service configuration mode.

After creating or modifying a profile, use the reset command in voice user-profile configuration mode to reset all phones on which this profile is downloaded to propagate the modifications.

Usage Guidelines

Use this command to modify the default value of the maximum video bandwidth for all video-capable phones associated with a Cisco Unified CME router. Default value is 1,000,000 kb/s.

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	This command can also be used for Cisco Unified SIP SRST.

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.

Usage Guidelines

This command supports Cisco Unified CME ephone hunt groups by allowing calls to be redirected more than the default number of times.

Usage Guidelines

This command sets the maximum number of concurrent presence subscriptions for both internal and external subscribe requests.

Usage Guidelines

With the default configuration, the Cisco UBE receives media packets from the inbound call leg, terminates them, and then reoriginates the media stream on an outbound call leg. Media flow-around enables media packets to be passed directly between the endpoints, without the intervention of the Cisco UBE. The Cisco UBE continues to handle routing and billing functions. Media flow-around for SIP-to-SIP calls is not supported.

Note	The Cisco UBE must be running Cisco IOS Release 12.3(1) or a later release to support media flow-around.

You can specify media flow-around for a voice class, all VoIP calls, or individual dial peers.

The transcoder high-density keyword can be enabled in any of the configuration modes with the same command format. If you are configuring the transcoder high-density keyword for dial peers, make sure that the media transcoder high-density command is configured on both the in and out legs.

The software does not support configuring the transcoder high-density keyword on any dial peer that is to handle video calls. The following scenarios are not supported:

• Dial peers used for video at any time. Configuring the media transcoder high-density command directly under the dial-peer or a voice-class media configuration is not supported.
• Dial peers configured on a Cisco UBE used for video calls at any time. The global configuration of the media transcoder high-density command under voice service voip is not supported.

To enable the media command on a Cisco 2900 or Cisco 3900 series Unified Border Element voice gateway, you must first enter the mode border-element command. This enables the media forking and media monitoring commands. Do not configure the mode border-element command on the Cisco 2800 or Cisco 3800 series platforms.

Usage Guidelines

All members configured under an ephone-hunt are initialized with HLogin. Use this command to initialize all non-shared static members to Hlogout.

Usage Guidelines

All members configured under a voice hunt-group are initialized with HLogin. Use this command to initialize all non-shared static members to Hlogout. If any member of a hunt group in a SIP phone logs out using the CLI command members logout , all other DN's of that phone in any hunt group are also logged out. This is because SIP phones only support phone level logout. For SCCP phones, only the DN that is configured with the CLI command members logout is logged out from the hunt group. Other member DN's do not logout as SCCP phones support line level logout.

Members Logout is not supported if the CLI command hunt-group logout DND is configured. Also, you cannot configure the CLI command members logout if the command list is configured.

Usage Guidelines

This command enables Cisco Unified CME to report missed calls on the specified phone. Use the all keyword to report missed calls to overlaid directory numbers. Only calls to an overlay set that are visibly presented on the phone are reported as missed calls. Calls to an overlay that are terminated by the caller before they are displayed on the phone are not reported as missed calls.

If the unique extension number for a phone is assigned to an overlay set on the phone, missed calls to that extension number are not reported unless you enable this command using the all keyword.

Usage Guidelines

The command allows an SCCP IP phone or an FXS analog phone to have its calls preempted if it is busy with lower precedence calls.

A phone with preemption disabled can still receive precedence calls in an MLPP network, but the phone itself does not get preempted. The preemption-disabled phone can be connected to a call that is preempted (at another device), in which case, that device receives preemption.

To apply a template to an SCCP phone, use the ephone-template command in ephone configuration mode.

Usage Guidelines

This command sets the MLPP domain type and number for the phone, and the maximum precedence level that a user can select when making MLPP calls from the phone.

The phone user can select a precedence level that is less than or equal to the value set with this command. Cisco Unified CME rejects the call if a user selects a precedence level that is higher than the level set with this command and the user receives an error tone.

If this command and the service-domain command are not enabled, the phone cannot make MLPP calls.

Emergency 911 calls are automatically assigned precedence level 0.

To apply a template to an SCCP phone, use the ephone-template command.

Usage Guidelines

This command enables the Mobility feature on the extension, which is required to enable the Single Number Reach (SNR) feature.

Usage Guidelines

Use the mobility command to enable a Cisco Unified SIP IP phone to receive calls on an extension, which is required to enable the Single Number Reach (SNR) feature.

Usage Guidelines

The mode command selects the SIP call control application to be used and enables associated configuration commands. By default, the router operates in SRST mode, allowing only phones failing over from Unified Communications Manager to register. Enter cme or cme-app modes to configure SIP phones and features for standalone call control use.

E-SRST mode provides a hybrid of these applications, where CME features may be configured to complement the SRST application.

For releases prior to Unified CME/SRST 11.7, the no form of this command clears only the mode-specific configurations (For example, source-address under voice register global configuration, and user credentials configured under voice register pool configuration). From Cisco IOS XE Everest 16.5.1 (Unified CME/SRST Release 11.7) onwards, the no form of this command clears all the voice register pools and voice register dns, along with mode-specific configurations.

Usage Guidelines

This command takes the specified live-feed audio stream and uses it as MOH for a Cisco Unified CME system. The connection for the live-feed audio stream is established as an automatically connected voice call. If the out-call keyword is used, the type of connection can include VoIP calls if voice activity detection (VAD) is disabled. The typical operation is for the MOH ephone-dn to establish a call to a local router E&M voice port.

Connection via E&M is the recommended mechanism because it requires minimal external components. The E&M port must be placed in 4-wire operation, using E&M immediate signaling and with the auto-cut-through option enabled. You directly connect a line-level audio feed (standard audio jack) to pins 3 and 6 of an E&M RJ-45 connector. The E&M WAN interface card (WIC) has a built-in audio transformer that provides appropriate electrical isolation for the external audio source. (The audio connection on the E&M port does not require loop current.) The signal immediate and auto-cut-through commands disable E&M signaling on this voice port. A G.711 audio packet stream is generated by the digital signal processor (DSP) on the E&M port.

If you are using an FXO voice port for live-feed MOH instead of an E&M port, connect the MOH source to the FXO voice port. This connection requires an external adapter to supply normal telephone company (telco) battery voltage with the correct polarity to the tip-and-ring leads of the FXO port. The adapter must also provide transformer-based isolation between the external audio source and the tip-and-ring leads of the FXO port.

Music from a live feed is continuously fed into the MOH playout buffer instead of being read from an audio file in flash memory. There is typically a two-second delay with live-feed MOH.

If the out-call keyword is used, an outbound call to the MOH live-feed source is attempted (or reattempted) every 30 seconds until the call is connected to the ephone-dn (extension) that has been configured for MOH. Note that this ephone-dn is not associated with any physical phone.

If the moh (ephone-dn) command is used without any keywords or arguments, the ephone-dn will accept an incoming call and use the audio stream from the call as the source for the MOH stream, displacing any audio stream that is available from a flash file. To accept an incoming call, the ephone-dn must have an extension or phone number configured for it. A typical usage would be for an external H.323-based server device to call the ephone-dn to deliver an audio stream to the Cisco CME system. Normally, only a single ephone-dn would be configured like this. If there is more than one ephone configured to accept incoming calls for MOH, the first ephone-dn that is successfully connected to a call (incoming or outgoing) is the MOH source for the system.

MOH can also be derived from an audio file when you use the moh command in telephony-service configuration mode with the filename argument. There can be only one MOH stream at a time in a Cisco CME system, and if both an audio file and a live feed have been specified for the MOH stream, the router seeks the live feed from the moh (ephone-dn) command first. If the live feed is found, the router displaces the audio file source. If the live feed is not found or fails at any time, the router falls back to the audio file source that was specified in the moh (telephony-service) command.

If you use the ip keyword to specify a multicast address in this command, the audio stream is sent to the multicast address in addition to serving as the MOH source. Additionally, if you specify a different multicast address using the multicast moh command under telephony-service configuration mode, the audio stream is also sent to the multicast address that you named in that command. It is therefore possible to send the live-feed audio stream to MOH and to two different multicast addresses: the one that is directly configured under the moh (ephone-dn) command and the one that is indirectly configured under the multicast moh (telephony-service) command.

A related command, the feed command, provides the ability to multicast an audio stream that is not the MOH audio stream.

Note	IP phones do not support multicast at 224.x.x.x addresses.

Usage Guidelines

This command enables MOH from .au and .wav format music files. MOH is played for G.711 callers and on-net VoIP and PSTN callers who are on hold in a Cisco CME system. Local callers within a Cisco CME system hear a repeating tone while they are on hold.

Audio files that are used for MOH must be copied to the Cisco CME router flash memory. A MOH file can be in .au or .wav file format; however, the file format must contain 8-bit 8-kHz data in A-law or mu-law data format. We recommend using a moh-file size greater than 100 KB.

If you want to replace or modify the audio file that is currently specified, you must first disable the MOH capability using the no moh command. The following example replaces file1 with file2:

Router(config-telephony)# moh file1
Router(config-telephony)# no moh
Router(config-telephony)# moh file2

If you specify a second file without first removing the original file, the MOH mechanism stops working and may require a router reboot to clear the problem.

A related command, the moh command in ephone-dn configuration mode, can be used to establish a MOH audio stream from a live feed. If you configure both commands, MOH falls back to playing music from the audio file if the live music feed is interrupted.

The multicast moh command allows you to use the MOH stream for a multicast broadcast.

When the multicast moh and debug ephone moh commands are both enabled, if you also use the no moh command, the debug output can be excessive and flood the console. Multicast MOH should be disabled before using the no moh command when the debug ephone moh command is enabled.

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	Music-on-hold files can be in .wav or .au file format; however, the file format must contain 8-bit 8-kHz data; for example, CCITT a-law or u-law data format.

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.

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	When live-feed is enabled there is no file caching for MOH-group 0.

Usage Guidelines

Use this command to assign a MOH group to a directory number in ephone-dn configuration mode. Use the number tag from 1 to 5 to specify the MOH group that you want to assign to this directory number.

Usage Guidelines

Normally, media packets (RTP packets) that are sent between IP phones in the same Cisco Unified CME system go directly to the other phone and do not travel through the Cisco Unified CME router. When these packets are sent from a remote IP phone to another IP phone in the same Cisco Unified CME system, they may be obstructed by a firewall. The mtp command instructs a phone to always send its media packets to the Cisco Unified CME router, which acts as a proxy and forwards the packets to the destination. Firewalls can then be easily configured to pass the RTP packets because the router uses a specified UDP port for media packets. In this way, RTP packets from remote IP phones can be delivered to IP phones on the same system. The default is that this function is off and that RTP packets that are sent from one IP phone to another IP phone in the same Cisco Unified CME system go directly to the other phone.

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

Usage Guidelines

Use the mtu command to define a value in bytes.. The mtu value ranges from 256 to 1406 bytes. The defaul value is 1290 bytes.

Usage Guidelines

This command enables multicast of the audio stream that is designated for MOH in a Cisco CME system. Use this command in voice moh-group configuration mode to enable multicast of audio stream for a specific MOH group.

A related command, the moh (ephone-dn) command, creates a MOH audio stream from an external live feed and optionally enables multicast on that stream. These two commands can be used concurrently to provide multicast of a live-feed MOH audio stream to two different multicast addresses.

Another related command, the feed command, enables multicast of an audio stream that is not the MOH audio stream.

When the multicast moh and debug ephone moh commands are both enabled, if you also use the no moh command, the debug output can be excessive and flood the console. Multicast MOH should be disabled before using the no moh command when the debug ephone moh command is enabled.

Note	IP phones do not support multicast at 224.x.x.x addresses.

Note	Multicast for live feed is not support in MOH groups.

Usage Guidelines

This command enables a Cisco Unified IP phone extension to receive MWI notification from an external voice-messaging system for all the Cisco Unified IP phones connected to the Cisco Unified CME router. This extension is a “dummy” extension and is not associated with any physical phone. The external voice-messaging system is able to communicate MWI status by making telephone calls to the dummy extension number, with the MWI information embedded in either the called or calling-party IP phone number.

This command cannot be used unless the number command is already configured for this extension (ephone-dn).

If you use an ephone-dn template to apply a command to an ephone-dn and you also use the same command in ephone-dn configuration mode for the same ephone-dn, the value that you set in ephone-dn configuration mode has priority.

Usage Guidelines

This command enables a particular extension on a SIP IP phone to receive MWI notification.

For Cisco Unified CME 4.1 and later versions, MWI requires that SIP phones must be configured with a directory number by using the number (voice register pool) command with the dn keyword; direct line numbers are not supported.

Usage Guidelines

Central voice-messaging servers that provide mailboxes for several Cisco Unified CME sites may use site codes or prefixes to distinguish among similarly numbered ranges of extensions at different sites. In Cisco Unified CME 4.0 and later versions, you can specify that your Cisco Unified CME system should accept unsolicited SIP Notify messages for message-waiting indication (MWI) that include a prefix string as a site identifier.

For example, an MWI message might indicate that the central mailbox number 5551234 has a voice message. In this example, the digits 555 are set as the prefix string or site identifier using the mwi prefix command. The local Cisco Unified CME system is able to convert 5551234 to 1234 and deliver the MWI to the correct phone. Without this prefix string manipulation, the system would reject an MWI indication for 5551234 as not matching the local Cisco Unified CME extension 1234.

Usage Guidelines

The transfer-system command must be used with the full-consult or full-blind keyword to enable H.450 call forwarding.

If you use an ephone-dn template to apply a command to an ephone-dn and you also use the same command in ephone-dn configuration mode for the same ephone-dn, the value that you set in ephone-dn configuration mode has priority.

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.

Usage Guidelines

Use this command to enable the Cisco CME 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.

Usage Guidelines

Use this command to subscribe an extension in a Cisco Unified CME router to receive MWI notification from a SIP-based MWI server, and use the mwi sip-server command to specify the IP address and port number for the external SIP-based MWI server. This function integrates a Cisco Unified CME router with a SIP-protocol-based MWI service.

If you use an ephone-dn template to apply a command to an ephone-dn and you also use the same command in ephone-dn configuration mode for the same ephone-dn, the value that you set in ephone-dn configuration mode has priority.

Usage Guidelines

Use this command to configure the IP address of an external SIP MWI server. This IP address is used with the mwi sip (ephone-dn) command to subscribe individual ephone-dn extension numbers to the notification list of the MWI SIP server. A SIP MWI client runs TCP by default.

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.

Central voice-messaging servers that provide mailboxes for several Cisco Unified CME sites may use site codes or prefixes to distinguish among similarly numbered ranges of extensions at different sites. In Cisco CME 3.2.3 and later versions, you can specify that your Cisco Unified CME system should accept unsolicited SIP Notify messages for message-waiting indication (MWI) that include a prefix string as a site identifier.

Usage Guidelines

This command designates a phone line other than the primary line to activate the MWI lamp on the phone. When a message is waiting for an ephone-dn associated with the designated line, the MWI lamp is turned on. When the message is heard, the MWI lamp is turned off. For phone lines other than the line that is designated to receive MWI, an envelope icon is displayed next to them when there is a message waiting.

Note that a logical phone “line” is not the same as a phone button. A line is a button that has one or more ephone-dns assigned to it. A button that has no ephone-dns assigned to it does not count as a line.

In most cases, one ephone-dn is assigned to one button on an ephone. When you set the mwi-line command to that button, the MWI lamp is turned on when there is a message waiting for that ephone-dn. When you set the mwi-line command to a button with a more complex configuration, the following rules apply:

• When a button has a single ephone-dn with primary and secondary numbers, the MWI lamp is turned on only when there is a message waiting for the primary number.
• When a button has several ephone-dns overlaid on it, the MWI lamp is turned on only when there is a message waiting for the first number in the list of ephone-dns.
• When a button is an overflow button for an overlay button, the MWI lamp is not turned on for any extension that might overflow to this button. If you set the mwi-line command to this button, the command is ignored.

Usage Guidelines

This command enables a directory number to receive audible, visual, or both audible and visual MWI notification from an external voice-messaging system. The external voice-messaging system is able to communicate MWI status by making telephone calls to the dummy extension, with the MWI information embedded in either the called or calling-party IP phone number.

Based on the capabilities of the IP phone and how the mwi-type command is configured, Message Waiting is communicated as follows:

• If the phone supports (visual) MWI and MWI is configured for the phone, Message Waiting light is lit.
• If the phone supports (visual) MWI only, Message Waiting light is lit regardless of the configuration.
• If the phone supports AMWI and AMWI is configured for the phone, stutter dial tone is sent to the phone when it goes off-hook.
• If the phone supports AMWI only and AMWI is configured, stutter dial tone is sent to the phone when it goes off hook regardless of the configuration.
• If a phone supports (visual) MWI and AMWI and both options are configured for the phone, the Message Waiting light is lit and a stutter dial tone is sent to the phone when it goes off-hook.

Before using this command:

• Create the directory number to be configured by using the number
• Enable MWI on this directory number by using the mwi command.

If you use an ephone-dn template to apply a command to a directory number and you also use the same command in ephone-dn configuration mode for the same number, the value that you set in ephone-dn configuration mode has priority.