Usage Guidelines

One label is allowed per extension (ephone-dn). The ephone-dn must already have a number that was set using the number command before a label can be created for it.

This command must be followed by a quick reboot of the phone on which the label appears, using the restart command.

Usage Guidelines

One label is allowed per extension (directory number). The directory number must already have a number that was set by using the number command before a label can be created for it.

After you configure this command, restart the phone by using the reset command.

Usage Guidelines

Use this command to create a list of member numbers for defining a hunt group.

List must contain 1 to 20 numbers.

A number cannot be added to a list unless it was already defined by using the number command.

Add or delete all numbers in a hunt group list at one time. You cannot add or single number to an existing list or remove one number from a list.

Any number in the list cannot be a pilot number of a parallel hunt group.

To allow dynamic membership in a hunt group, use asterisks to represent wildcard slots in the list command. To allow an ephone-dn to use one of the wildcard slots to dynamically join a hunt group, use the ephone-hunt login command under that ephone-dn. Ephone-dns are disallowed from joining hunt groups by default, so you have to explicitly allow this behavior for each ephone-dn that you want to be able to log into hunt groups.

The show ephone-hunt command displays the numbers associated to ephone-dns that are joined to groups at the time that the command is run, in addition to static members of the hunt group. Static hunt group members are the numbers that are explicitly named in the list command.

Usage Guidelines

This command creates the list of numbers to include in a voice hunt-group. Phones with these numbers ring when the hunt group pilot number is dialed. The numbers must be assigned to directory numbers with the number command.

All numbers in a hunt group list are added or deleted at one time. You cannot add a number to an existing list or remove a number from a list.

The pilot number of a parallel hunt group and shared-line numbers are not supported.

A phone number associated with an FXO port is not supported in parallel hunt groups.

Usage Guidelines

This command specifies the telephone number that is speed-dialed to access the Live Record feature when the LiveRcd soft key on a Cisco Unified IP phone is pressed. This telephone number is used for all Cisco Unified IP phones connected to the router.

This telephone number must match the Live Record number configured in Cisco Unity Express.

Usage Guidelines

This command updates the Cisco Unified CME configuration file for the specified type of Cisco Unified IP phone to add the name of the firmware file to be loaded by a particular phone type. The firmware filename also provides the version number for the phone firmware that is in the file. When a phone is started up or rebooted, the phone reads the configuration file to determine which firmware file it must load and then looks for that firmware file on the TFTP server.

If applicable, Cisco Unified IP phones update themselves with new phone firmware whenever they are started up or rebooted.

A separate load command is needed for each type of phone. The Cisco Unified IP Phones 7940 and 7940G and the Cisco Unified IP Phones 7960 and 7960G have the same phone firmware and share the 7960-7940 keyword.

Before Cisco Unified CME 7.0(1):

• Do not include the file suffix (.bin, .sbin, .loads) for any phone type except Cisco ATA and Cisco Unified IP Phone 7905 and 7912 when you configure the load command in telephony-service configuration mode. For example:

Router(config-telephony)# load 7941 SCCP41.8-2-2SR2S
Router(config-telephony)# 

• You must also configure the tftp-server command to enable TFTP access to the firmware files by Cisco Unified IP phones.

In Cisco Unified CME 7.0(1) and later versions:

• When specifying the load command for phone firmware versions later than version 8-2-2 for all phone types and you use the file suffix in the filename, the tftp-server bindings are automatically added for all the files forwarded for that load. For example:

Router(config-telephony)# load 7941 SCCP41.8-3-3S.loads
Router(config-telephony)# 

• The load command is enhanced to automatically create TFTP bindings for phone firmware files if the cnf-file location command is configured with the flash or slot0 keyword. You are no longer required to configure the tftp-server command to create TFTP bindings only if the location of the cnf files is router flash or slot 0 memory. If the cnf-file location command is configured for something other than flash or slot 0, such as a TFTP server (url) or system memory (system:its/), you must still configure the tftp-server command to create TFTP bindings for phone firmware files. Use the complete filename, including the file suffix, when you configure the tftp-server command for phone firmware versions later than version 8-2-2 for all phone types.

To verify TFTP bindings, including the dictionary, language, and tone configuration files that are associated with the ISO-3166 codes that have been selected, use the show telephony-service tftp-bindings command.

After associating a firmware file with a Cisco Unified IP phone, use the reset command to reboot the phone.

Usage Guidelines

This command updates the Cisco Unified CME configuration file for the specified type of IP phone to add the name of the correct firmware file that the phone should load. This filename also provides the version number for the phone firmware that is in the file. Later, whenever a phone is started up or rebooted, the phone reads the configuration file to determine the name of the firmware file that it should load and then looks for that firmware file on the TFTP server.

A separate load command is needed for each type of phone. The Cisco Unified IP Phone 7940 and 7940G and Cisco Unified IP Phone 7960 and 7960G have the same phone firmware and share the 7960-7940 keyword. The Cisco Unified IP Phone 3911 and Cisco Unified IP Phone 3951 have the same phone firmware and share the 3951 keyword.

For certain IP phones, such as the Cisco Unified IP Phone 7906G, 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971G, there are multiple firmware files. For these phones, use the TERMnn.x-y-x-w.loads or SIPnn.x-y-x-w.loads firmware filename for the load command, without the .loads file extension. For these phones, you do not configure the load command for any firmware file other than the TERM.loads or SIP.loads firmware file.

Following the load command, use the tftp-server command to enable TFTP access to the file by Cisco Unified IP phones. The file extension is required when using the tftp-server command.

The load command must be followed by a reboot of the phones. Plug in a new IP phone or use the reset command to reboot an IP phone that is already connected to the Cisco router.

Usage Guidelines

This command is used with Cisco Unified CME phone authentication to sign configuration files that are not created by Cisco Unified CME. This command also loads the signed and unsigned versions of the file on the TFTP server. To simply serve an already signed file on the TFTP server, use this command without the sign and create keywords.

The create keyword should be used with the sign keyword the first time that this command is used for each file. The create keyword is not maintained in the running configuration; this prevents signed files from being recreated during every reload.

Usage Guidelines

This command specifies the G.711 a-law or u-law 8-KHz encoded audio file (.wav or .au format) for the announcement that plays to callers when the call leaves the Cisco Unified CME router on the trunk or when the user places a call to a different domain.

The mlpp indication command must be enabled (default) for a phone to play precedence announcements.

This command is not supported by Cisco IOS help. If you type ? , Cisco IOS help does not display a list of valid entries.

Usage Guidelines

Use this command to create 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 public safety answering poins (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.

Usage Guidelines

The local password is used to authenticate XML API requests on the network management server. If the password is not set, an XML API query fails local authentication.

The password string is stored as plain text. No encryption is supported.

Usage Guidelines

Cisco Unified CME captures and time-stamps events, such as phones registering and unregistering and extension status, and stores them in an internal buffer. This command sets the maximum number of events, or entries, that can be stored in the table. One event equals one entry. The retain-timer keyword sets the number of minutes that events are kept in the buffer before they are deleted.

The event table can be viewed using the show fb-its-log command.

Usage Guidelines

Use this command to enable 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 optional command is enabled by default.

Usage Guidelines

This command defines the after-hours login timer. Individual users on specified phones can override call blocking by logging in using a personal identification number (PIN). The after-hours login timer deactivates user login on all phones at a specific time and deactivates a login session automatically after a phone is idle for a specified period of time.

The login command applies only to IP phones that have soft keys, such as the Cisco Unified IP Phone 7940 and 7940G and the Cisco Unified IP Phone 7960 and 7960G.

For this command to take effect, fast reboot and reregister all phones in Cisco Unified CME by using the restart all command in telephony-service configuration mode.

When a Cisco Unified CME router is rebooted, the login status for all phones is reset to the default.

Usage Guidelines

Use this command to define the URL for the file to be used by SIP phones connected in Cisco Unified CME. The file that is displayed must be encoded in eXtensible Markup Language (XML) by using the Cisco XML document type definition (DTD). For more information about Cisco DTD formats, see the Cisco IP Phone Services Application Development Notes.

After you configure this command, restart the phones by using the reset command.

Usage Guidelines

Use this command in ephone configuration mode to enable a supported IP phone registered in Cisco Unified CME for extension mobility and to apply a default logout profile to the ephone being configured.

In Cisco Unified CME 4.2, extension mobility is supported only on SCCP IP phones.

In Cisco Unified CME 8.6 extension mobility is supported on SIP phones.

Extension mobility is not supported on non-display IP phones.

Extension mobility is not supported for analog devices.

Before using this command, you must create a logout profile to be applied to this phone by using the voice logout-profile command.

You cannot apply more than one logout profile to an ephone. If you attempt to apply a second logout profile to an ephone to which a profile has already been applied, the second profile will overwrite the first logout profile configuration.

Usage Guidelines

The loopback-dn command is used to configure two ephone-dn virtual voice ports as back-to-back-connected voice-port pairs. A call presented on one side of the loopback-dn pair is reoriginated as a new call on the opposite side of the loopback-dn pair. The forward , strip, prefix , and suffix keywords can be used to manipulate the original called number that is presented to the incoming side of the loopback-dn pair to generate a modified called number to use when reoriginating the call at the opposite side of the loopback-dn pair. For loopback-dn configurations, you must always configure ephone-dn virtual voice ports as cross-coupled pairs.

Note

Use of loopback-dn configurations within a VoIP network should be restricted to resolving critical network interoperability service problems that cannot otherwise be solved. Loopback-dn configurations are intended to be used in VoIP network interworking situations in which the only other alternative would be to make use of back-to-back-connected physical voice ports. Loopback-dn configurations emulate the effect of a back-to-back physical voice-port arrangement without the expense of the physical voice-port hardware. A disadvantage of loopback-dn configurations is that, because digital signal processors (DSPs) are not involved in a loopback-dn arrangement, the configuration does not support interworking or transcoding between calls that use different voice codecs. In many cases, the use of back-to-back physical voice ports that do use DSPs to resolve VoIP network interworking issues is preferred, because it introduces fewer restrictions in terms of supported codecs and call flows. Also, loopback-dns do not support T.38 fax relay.

Note

We recommend that you create the basic ephone-dn configuration for both ephone-dn entries before configuring the loopback-dn option under each ephone-dn. The loopback-dn mechanism should be used only in situations where the voice call parameters for the calls on either side of the loopback-dn use compatible configurations; for example, compatible voice codec and dual tone multifrequency (DTMF) relay parameters. Loopback-dn configurations should be used only for G.711 voice calls.

The loopback-dn arrangement allows an incoming telephone call to be terminated on one side of the loopback-dn port pair and a new pass-through outgoing call to be originated on the other side of the loopback-dn port pair. The loopback-dn port pair normally works with direct cross-coupling of their call states; the alerting call state on the outbound call segment is associated with the ringing state on the inbound call segment.

The loopback-dn mechanism allows for call operations (such as call transfer and call forward) that are invoked for the call segment on one side of the loopback-dn port pair to be isolated from the call segment that is present on the opposite side of the loopback-dn port pair. This approach is useful when the endpoint devices associated with the two different sides have mismatched call-transfer and call-forwarding capabilities. The loopback-dn arrangement allows for call-transfer and call-forward requests to be serviced on one side of the loopback-dn port pair by creating hairpin-routed calls when necessary. The loopback-dn arrangement avoids the propagation of call-transfer and call-forward requests to endpoint devices that do not support these functions.

The loopback-dn command provides options for controlling the called-number digits that are passed through from the incoming side to the outgoing side. The available digits can be manipulated with the forward , strip , prefix , and suffix keywords.

The forward keyword defines the number of digits in the original called number to forward to the other ephone-dn in the loopback-dn pair. The default is set to forward all digits. The strip keyword defines the number of leading digits to be stripped from the original called number before forwarding to the other ephone-dn in the loopback-dn pair. The default is set to not strip any digits. The forward and strip commands are mutually exclusive and can be used with any combination of the prefix and suffix keywords.

The prefix keyword defines a string of digits to add in front of the forwarded number.

The suffix keyword is most commonly used to add a terminating “#” (pound-sign) character to the end of the forwarded number to indicate that no more digits should be expected. The pound-sign character indicates to the call-routing mechanism that is processing the forwarded number that the forwarded number is complete. Providing an explicit end-of-number character also avoids a situation in which the call-processing mechanism waits for the interdigit timeout period to expire before routing the call onward using the forwarded number.

Note	The Cisco IOS command-line interface (CLI) requires that arguments with character strings that start with the pound-sign (#) character be enclosed within quotation marks; for example, “#”.

The retry keyword is used to suppress a far-end busy indication on the outbound call segment. Instead of returning a busy signal to the call originator (on the incoming call segment), a loopback-dn presents an alerting or ringing tone to the caller and then periodically retries the call to the final far-end destination (on the outgoing call segment). This is not bidirectional. To prevent calls from being routed into the idle outgoing side of the loopback-dn port pair during the idle interval that occurs between successive outgoing call attempts, configure the outgoing side of the loopback-dn without a number so that there is no number to match for the inbound call.

The auto-con keyword is used to configure a premature trigger for a connected state for an incoming call segment while the outgoing call segment is still in the alerting state. This setup forces the voice path to open for the incoming call segment and support the generation of in-band call progress tones for busy, alerting, or ringback. The disadvantage of the auto-con keyword is premature opening of the voice path during the alerting stage and also triggering of the beginning of billing for the call before the call has been answered by the far end. These disadvantages should be considered carefully before you use the auto-con keyword.

The codec keyword is used to explicitly select the A-law or mu-law type of G.711 and to provide A-law to mu-law conversion if needed. Setting the codec type on one side of the loopback-dn forces the selection of A-law or mu-law for voice packets that are transmitted from that side of the loopback-dn. To force the A-law or mu-law G.711 codec type for both voice packet directions, set the codec type on both sides of the loopback-dn. Loopback-dn configurations are used only with G.711 calls. Other voice codec types are not supported.

Usage Guidelines

Analog Phones

An incoming call to an analog phone is associated with the LPCOR policy specified with this command in the voice port. Otherwise the LPCOR policy specified in the trunk group is used.

SCCP IP Phones (Local or Remote)

An incoming call to an SCCP IP phone is associated with the LPCOR policy specified with this command in ephone or ephone template configuration mode. The ephone configuration has precedence over the ephone-template configuration. All directory numbers on the phone share the same LPCOR setting.

SIP IP Phones (Local or Remote)

An incoming call to a SIP IP phone is associated with the LPCOR policy specified with this command in voice register pool or voice register template configuration mode. The voice register pool configuration has precedence over the voice register template configuration. All directory numbers on the phone share the same LPCOR setting.

Note	This command is not supported for phones configured with the lpcor type mobility command.

• Phones that share a directory number must be configured with the same LPCOR policy. Different LPCOR settings on shared-line phones are not supported.

PSTN Trunks

An incoming call to the PSTN is associated with the LPCOR policy specified with this command in the voice port. Otherwise the LPCOR policy specified in the trunk group is used. The voice port configuration takes precedence.

VoIP Trunks (H.323 or SIP)

An incoming call to a VoIP trunk is associated with the LPCOR policy specified with this command in voice service configuration mode if the remote IP address is not found in the IP trunk subnet table created with the voice lpcor ip-trunk subnet incoming command.

Usage Guidelines

Analog Phones

An outgoing call from an analog phone is associated with the LPCOR policy specified with this command in the voice port. Otherwise the LPCOR policy specified in the trunk group is used.

SCCP IP Phones (Local or Remote)

An outgoing call from an SCCP IP phone is associated with the LPCOR policy specified with this command in ephone configuration or ephone template configuration mode. The ephone configuration has precedence over the ephone-template configuration. All directory numbers on the phone share the same LPCOR setting.

SIP IP Phones (Local or Remote)

An outgoing call from a SIP IP phone is associated with the LPCOR policy specified with this command in voice register pool or voice register template configuration mode. The voice register pool configuration has precedence over the voice register template configuration. All directory numbers on the phone share the same LPCOR setting.

Note	This command is not supported for phones configured with the lpcor type mobility command.

• Phones that share a directory number must be configured with the same LPCOR policy. Different LPCOR settings on shared-line phones are not supported.

PSTN Trunks

An outgoing call from the PSTN uses the LPCOR policy specified with this command in the voice port if the outbound dial peer is configured with the port command. Otherwise the outgoing call uses the LPCOR policy specified with this command in the trunk group if the outbound dial peer is configured with the trunkgroup command.

VoIP Trunks (H.323 or SIP)

An outgoing VoIP call uses the LPCOR policy specified with this command in the outbound dial peer. Otherwise the outgoing call uses the default LPCOR policy.

Usage Guidelines

According to the Telecom Regulatory Authority of India (TRAI) requirements, an IP phone can accept both PSTN and VoIP calls if it is locally registered to Cisco Unified CME through the LAN. Select the local keyword for this type of phone.

If an IP phone is registered remotely to Cisco Unified CME through the WAN, PSTN calls must be blocked from that remote IP phone. Select the remote keyword for this type of phone.

A static LPCOR policy is applied to an IP phone if the phone registers to Cisco Unified CME from the same region (local or remote) permanently.

If an IP phone moves between the local and remote regions, such as an Extension Mobility phone, Cisco IP Communicator softphone, or remote teleworker, select the mobile keyword. The LPCOR policy is assigned dynamically based on the phone’s currently registered IP address.

If you use a phone template to apply a command to a phone and you also use the same command in the phone configuration of the same phone, the value in phone configuration has priority.