To specify the audio file used for the isolated code announcement, use the ica command in voice MLPP configuration mode. To disable use of this audio file, use the no form of this command.

Syntax Description

Command Default

No announcement is played.

Command Modes

Voice MLPP configuration (config-voice-mlpp)

Command History

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 service or equipment problems prevent completion of their call.

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.

Related Commands

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.

Syntax Description

Command Default

No SIP IP phone is configured.

Command Modes

Voice register pool configuration (config-register-pool)

Command History

Usage Guidelines

Configure this command before configuring any other command in voice register pool configuration mode.

This 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:

• Verification of the local Layer 2 MAC address using the router’s Address Resolution Protocol (ARP) cache.
• Verification of the known single static IP address (or DHCP dynamic IP address within a specific subnet) of the Cisco SIP IP phone.

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	For Cisco Unified SIP SRST, this command also allows explicit identification of locally available set of Cisco SIP IP phones.

Related Commands

To import a trusted certificate in PEM format from flash memory to the CTL file of an IP phone, use the import certificate command in ctl-client configuration mode. To return to the default, use the no form of this command.

Syntax Description

Command Default

None

Command Modes

CTL-client configuration (config-ctl-client)

Command History

Usage Guidelines

A CTLFile.tlv file should appear in the flash location after using the regenerate command in ctl-client configuration mode. If the file is missing, use the debug ctl-client command, followed by the regenerate command.

Related Commands

To add a logical partitioning class of restriction (LPCOR) group to the IP-phone subnet table, use the index command in LPCOR ip-phone subnet configuration mode. To remove a resource, use the no form of this command.

Syntax Description

Command Default

No index entry is configured.

Command Modes

LPCOR ip-phone subnet configuration (cfg-lpcor-ipphone-subnet)

Command History

Usage Guidelines

This command is used for mobility-type phones only, which can include Extension Mobility phones, teleworker remote phones, and Cisco IP Communicator softphones.

Two IP-phone subnet tables, containing up to 50 index entries, can be defined on each Cisco Unified CME router. One table is for incoming calls and the other table is for outgoing calls.

A LPCOR policy is dynamically associated with calls to and from a mobility-type phone by matching its current IP address or DHCP pool in the IP-phone subnet table. If the LPCOR policy cannot be provisioned from the IP-phone subnet table, the default LPCOR policy for mobility-type phones is used.

Entries in the IP-phone subnet tables are indexed in ascending order. The lookup of entries is in sequential ascending order. After Cisco Unified CME finds a matching entry, the corresponding LPCOR policy is associated with the call. Even if there are other entries that are a better match, only the first match is used.

For instance, in the example below, if a call originates from an IP phone with IP address 10.1.10.3, LPCOR policy local_g4 is associated with the incoming call instead of LPCOR policy local_g5 even though local_g5 is a better match.

Related Commands

To add a logical partitioning class of restriction (LPCOR) resource group to the IP trunk subnet table, use the index command in LPCOR IP-trunk subnet configuration mode. To remove a resource, use the no form of this command.

Syntax Description

Command Default

No index entry is configured.

Command Modes

LPCOR IP-trunk subnet configuration (cfg-lpcor-iptrunk-subnet)

Command History

Usage Guidelines

One IP-trunk subnet table, containing up to 50 index entries, can be defined on each Cisco Unified CME router for incoming VoIP trunk calls (H.323 or SIP).

An incoming VoIP trunk call is associated with a LPCOR policy by matching the remote IP address to an entry in the incoming IP-trunk subnet table. If that is not successful, the LPCOR policy in voice service configuration mode is applied.

Entries in the IP-trunk subnet table are indexed in ascending order. The lookup of entries is in sequential ascending order. After Cisco Unified CME finds a matching entry, it associates the corresponding LPCOR policy with the call. Even if there are other entries that are a better match, only the first match is used.

In the following example, an incoming VoIP call with a remote IP address of 172.19.22.25 is associated with sip_group1 even though voip_group2 is a better match.

Related Commands

To create an intercom by programming a pair of extensions (ephone-dns) to automatically call and answer each other, use the intercom command in ephone-dn configuration mode. To remove an intercom, use the no form of this command.

Syntax Description

Command Default

Intercom functionality is disabled.

Command Modes

Ephone-dn configuration (config-ephone-dn)

Command History

Usage Guidelines

This command is used to dedicate a pair of Cisco ephone-dns for use as a “press to talk” two-way intercom between Cisco IP phones. Intercom lines cannot be used in shared-line configurations. If an ephone-dn is configured for intercom operation, it must be associated with one Cisco IP phone only. The intercom attribute causes an IP extension (ephone-dn) to operate in autodial fashion for outbound calls and autoanswer-with-mute for inbound calls.

The barge-in keyword allows inbound intercom calls to force an existing call on the called phone into the call-hold state to allow the intercom call to be answered immediately. The no-auto-answer keyword creates for the IP phone line a connection that resembles a private line, automatic ringdown (PLAR). The label keyword defines a text label for the intercom.

Following this command, the intercom ephone-dns are assigned to ephones using the button command. Following the button command, the restart command must be used to initiate a quick reboot of the phones to which this intercom is assigned.

The default intercom command behavior is speakers are set to mute automatically when phones receive intercom calls. For example, if phone user 1 places an intercom call and connects to phone user 2, user 2 will hear user 1, but user 1 will not hear user 2. To be heard, user 2 must first disable the speaker’s mute function. The benefit is people who receive intercom calls can use the mute button to control when they will be heard initially.

The no-mute keyword deactivates the speaker mute function when IP phones receive intercom calls. For example, if phone user 1 makes an intercom call to phone user 2, both users will hear each other upon connection. The benefit is that people who receive intercom calls do not have to disable their speaker’s mute function to be heard, but their conversations and nearby background sounds will be heard the moment an intercom call to them is connected—regardless of whether they are ready to take a call or not.

The intercom command allows you to add a paging number to behave as a push-to-talk (ptt) feature. More information on the push-to-talk feature is available at this link: http:/​/​www.cisco.com/​en/​US/​docs/​voice_ip_comm/​cucme/​admin/​configuration/​guide/​cmelabel.html#wpmkr1048855

Related Commands

To enable the intercom call option on a Cisco Unified SIP IP phone, use the intercom command in voice register dn configuration mode. To prevent a Cisco Unified SIP IP phone from making an intercom call, use the no form of this command.

Syntax Description

Command Default

The Cisco Unified SIP IP phone cannot make or receive an intercom call.

Command Modes

Voice register dn configuration (config-register-dn)

Command History

Usage Guidelines

The intercom line cannot be the primary line of a Cisco Unified SIP IP phone and cannot be shared among Cisco Unified SIP IP phones.

When the intercom speed-dial option is not configured, the intercom line waits for the user to dial the destination number.

Related Commands

To assign an MOH group for calls from an internal directory number, use the internal-call command in telephony-service configuration mode. To disable the internal-call command, use the no form of this command.

Syntax Description

Command Default

No internal-call is configured.

Command Modes

Telephony-service configuration (config-telephony-service)

Command History

Usage Guidelines

Before using this command make sure you have MOH-groups configured under voice-moh-group configuration mode. This command allows you to assign a MOH-group for all calls from an internal directory number. MOH group tag identifies the unique number assigned to a MOH group. Range for MOH group tag is from 0 to 5, where 0 represents MOH configuration in telephony service.

Related Commands

To enable ip address trusted authentication for incoming VoIP (H.323/SIP) calls, use the ip address trusted authenticate command in voice service voip mode. To disable ip address trusted authentication, use the no form of this command.

Syntax Description

This command has no arguments or keywords.

Command Default

IP address trusted list authenticate is enabled.

Command Modes

Voice Service Voip

Command History

Usage Guidelines

Use this command to enable the ip address trusted authentication for incoming H.323 or SIP trunk calls for toll fraud prevention on Cisco Unified CME.

Related Commands

To issues a cause-code when the incoming call is rejected by the IP address trusted authentication, use the ip address trusted call-block cause command in voice service voip mode. To stop the IP address trusted authentication process from sending a call-block cause, use the no form of this command.

Syntax Description

Command Default

A call-reject (21) cause-code is issued to disconnect the incoming VoIP calls.

Command Modes

Voice Service voip.

Command History

Usage Guidelines

Use this command to issue a cause-code when the incoming call is rejected by the IP address trusted authentication. You can issue a specific call-block cause code using any one of the Q.850 call reject cause codes.

Related Commands

To manually add multiple IP addresses for incoming VoIP (H.323/SIP) calls, use the ip address trusted list command in voice service voip mode. To turn off the list, use the no form of this command.

Syntax Description

Command Default

IP address trusted list is disabled.

Command Modes

Voice Service Voip.

Command History

Usage Guidelines

Use this command to manually add unique and multiple IP addresses to a list of trusted IP addresses. You can add up to 100 IPv4 addresses in the ip address trusted list. No duplicate IP addresses are allowed.

Related Commands

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 telephony-service or voice register global configuration mode. To reset to the default value, use the no form of this command.

Syntax Description

Command Default

DSCP for media is ef. DSCP for service is 0. DSCP for signaling is cs3. DSCP for video is af41.

Command Modes

Telephony-service configuration (config-telephony)
Voice register global configuration (config-register-global)

Command History

Usage Guidelines

This command allows you to set different priority levels for different types of network traffic sent by the Cisco Unified CME router. Differentiated Services is a method of prioritizing specific network traffic based on the QoS specified by each packet. You can set different DSCP values, for example, for video and audio streams.

Cisco Unified CME downloads the configured DSCP value to the phones in their configuration files and all control messages and RTP streams are marked with the preferred DSCP value. Use this command in telephony-service mode to set the DSCP for SCCP phones. Use the command in voice register global mode to set the value for SIP 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.

Related Commands

To enable the Cisco Unified CME or Cisco Unified 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.

Syntax Description

Command Default

Default port number in Cisco Unified CME is 2444. Default port number in Cisco Unified SRST is 2445.

Command Modes

Credentials configuration (config-credentials)

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.

Related Commands

To identify the IP address and port through which IP phones communicate with a Cisco Unified CME router, use the ip source-address command in telephony-service or group configuration mode. To disable the router from receiving messages from Cisco Unified IP phones, use the no form of this command.

Syntax Description

Command Default

The IP address for communicating with phones is not defined.

Command Modes

Telephony-service configuration (config-telephony)
Group configuration (conf-tele-group)

Command History

Usage Guidelines

This command enables a router to receive messages from Cisco Unified IP phones through the specified IP address and port.

The Cisco Unified CME router cannot communicate with Cisco Unified CME phones if the IP address of the port to which they are attached is not configured. In Cisco Unified CME 8.0 and later versions, the Cisco Unified CME router can receive messages from IPv6-enabled or IPv4-enabled IP phones or from phones in dual-stack (both IPv6 and IPv4) mode.

• In Cisco Unified CME 8.0 and later versions: If the IP phones connected to Cisco Unified CME were configured for dual-stack mode by using dual-stack keyword with the protocol mode command, configure this command with the IPv6 address.
• In Cisco Unified CME 8.0 and later versions: If the IP phones to be connected to the port to be configured are IPv4-enabled only or IPv6-enabled only, configure this command with the corresponding IPv4 or IPv6 address.

For IPv6: Do not configure the port port keyword argument combination in this command to change the value from the default (2000). If you change the port number, IPv6 CEF packet switching engine will not be able to handle the IPv6 SCCP phones and various packet handling problems may occur when more than a dozen (approximately) calls in IPv6 are going on.

Use the strict-match keyword to instruct the router to reject IP phone registration attempts if the IP server address used by the phone does not match the source address.

Prior to Cisco IOS Telephony Services (Cisco ITS) V2.1, this command helped the router to autogenerate the SEPDEFAULT.cnf file, which was stored in the flash memory of the router. The SEPDEFAULT.cnf file contains the IP address of one of the Ethernet ports of the router to which the phone should register.

In ITS V2.1 and in Cisco CME 3.0 and later versions, the configuration files were moved to system:/its/. The file named Flash:SEPDEFAULT.cnf that was used with previous Cisco ITS versions is obsolete, but is retained as system:/its/SEPDEFAULT.cnf to support upgrades from older phone firmware.

For systems using Cisco ITS V2.1 or later versions, the IP phones receive their initial configuration information and phone firmware from the TFTP server associated with the router. In most cases, the phones obtain the IP address of their TFTP server using the option 150 command and Dynamic Host Configuration Protocol (DHCP). For Cisco ITS or Cisco CME operation, the TFTP server address obtained by the Cisco Unified IP phones should point to the router IP address. The Cisco IP phones attempt to transfer a configuration file called XmlDefault.cnf.xml. This file is automatically generated by the router through the ip source-address command and is placed in router memory. The XmlDefault.cnf.xml file contains the IP address that the phones use to register for service, using the SCCP. This IP address should correspond to a valid Cisco CME router IP address (and may be the same as the router TFTP server address).

Similarly, when an analog telephone adapter (ATA) such as the ATA-186 is attached to the Cisco Unified CME router, the ATA receives very basic configuration information and firmware from the TFTP server XmlDefault.cnf.xml file. The XmlDefault.cnf.xml file is automatically generated by the Cisco Unified CME router with the ip source-address command and is placed in the router’s flash memory.

By specifying a second Cisco Unified CME router in the ip source-address command, you improve the failover time for phones.

Related Commands