Configuring Phones to Make Basic Calls

This chapter describes how to configure Cisco Unified IP phones in Cisco Unified Communications Manager Express (Cisco Unified CME) so that you can make and receive basic calls.


Caution

The Interactive Voice Response (IVR) media prompts feature is only available on the IAD2435 when running IOS version 15.0(1)M or later.

Prerequisites for Configuring Phones to Make Basic Calls

Restrictions for Configuring Phones to Make Basic Calls

When you are configuring dial peers or ephone-dns, including park slots and conferencing extensions, on Cisco Integrated Services Router Voice Bundles, the following message may appear to warn you that free memory is not available: 

%DIALPEER_DB-3-ADDPEER_MEM_THRESHOLD: Addition of dial-peers limited by available memory

To configure more dial peers or ephone-dns, increase the DRAM in the system. A moderately complex configuration may exceed the default 256 MB DRAM and require 512 MB DRAM. Note that many factors contribute to memory usage, in addition to the number of dial peers and ephone-dns configured.

Information About Configuring Phones to Make Basic Calls

Phones in Cisco Unified CME

An ephone, or “Ethernet phone,” for SCCP or a voice-register pool for SIP is the software configuration for a phone in Cisco Unified CME. This phone can be either a Cisco Unified IP phone or an analog phone. Each physical phone in your system must be configured as an ephone or voice-register pool on the Cisco Unified CME router to receive support in the LAN environment. Each phone has a unique tag, or sequence number, to identify it during configuration.

For information on the phones supported in Cisco Unified CME Release 8.8 and later versions, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified Secure SRST.

Directory Numbers

A directory number, also known as an ephone-dn for SCCP or a voice-register dn for SIP, is the software configuration in Cisco Unified CME that represents the line connecting a voice channel to a phone. A directory number has one or more extension or telephone numbers associated with it to allow call connections to be made. Generally, a directory number is equivalent to a phone line, but not always. There are several types of directory numbers, which have different characteristics.

Each directory number has a unique dn-tag , or sequence number, to identify it during configuration. Directory numbers are assigned to line buttons on phones during configuration.

One virtual voice port and one or more dial peers are automatically created for each directory number, depending on the configuration for SCCP phones, or for SIP phones, when the phone registers in Cisco Unified CME.

Because each directory number represents a virtual voice port in the router, the number of directory numbers that you create corresponds to the number of simultaneous calls that you can have. This means that if you want more than one call to the same number to be answered simultaneously, you need multiple directory numbers with the same destination number pattern.

The directory number is the basic building block of a Cisco Unified CME system. Six different types of directory numbers can be combined in different ways for different call coverage situations. Each type will help with a particular type of limitation or call-coverage need. For example, if you want to keep the number of directory numbers low and provide service to a large number of people, you might use shared directory numbers. Or if you have a limited quantity of extension numbers that you can use and you need to have a large quantity of simultaneous calls, you might create two or more directory numbers with the same number. The key is knowing how each type of directory number works and its advantages.

Not all types of directory numbers can be configured for all phones or for all protocols. In the remaining information about directory numbers, we have used SCCP in the examples presented but that does not imply exclusivity. The following sections describe the types of directory numbers in a Cisco Unified CME system:

Single-Line

A single-line directory number has the following characteristics:

  • Makes one call connection at a time using one phone line button. A single-line directory number has one telephone number associated with it.

  • Should be used when phone buttons have a one-to-one correspondence to the PSTN lines that come into a Cisco Unified CME system.

  • Should be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI), loopback, and music-on-hold (MOH) feed sources.

  • Must have more than one single-line directory number on a phone when used with multiple-line features like call waiting, call transfer, and conferencing.

  • Can be combined with dual-line directory numbers on the same phone.


Note

You must make the choice to configure each directory number in your system as either dual-line or single-line when you initially create configuration entries. If you need to change from single-line to dual-line later, you must delete the configuration for the directory number, then recreate it.

Single-Line Directory Number shows a single-line directory number for an SCCP phone in Cisco Unified CME.

Figure 1. Single-Line Directory Number

Dual-Line

A dual-line directory number has the following characteristics:

  • Has one voice port with two channels.

  • Supported on IP phones that are running SCCP; not supported on IP phones that are running SIP.

  • Can make two call connections at the same time using one phone line button. A dual-line directory number has two channels for separate call connections.

  • Can have one number or two numbers (primary and secondary) associated with it.

  • Should be used for a directory number that needs to use one line button for features like call waiting, call transfer, or conferencing.

  • Cannot be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI), loopback, and music-on-hold (MOH) feed sources.

  • Can be combined with single-line directory numbers on the same phone.


Note

You must make the choice to configure each directory number in your system as either dual-line or single-line when you initially create configuration entries. If you need to change from single-line to dual-line later, you must delete the configuration for the directory number, then recreate it.

Dual-Line Directory Number shows a dual-line directory number for an SCCP phone in Cisco Unified CME.

Figure 2. Dual-Line Directory Number

Octo-Line

An octo-line directory number supports up to eight active calls, both incoming and outgoing, on a single button of a SCCP phone. Unlike a dual-line directory number, which is shared exclusively among phones (after a call is answered, that phone owns both channels of the dual-line directory number), an octo-line directory number can split its channels among other phones that share the directory number. All phones are allowed to initiate or receive calls on the idle channels of the shared octo-line directory number.

Because octo-line directory numbers do not require a different ephone-dn for each active call, one octo-line directory number can handle multiple calls. Multiple incoming calls to an octo-line directory number ring simultaneously. After a phone answers a call, the ringing stops on that phone and the call-waiting tone plays for the other incoming calls. When phones share an octo-line directory number, incoming calls ring on phones without active calls and these phones can answer any of the ringing calls. Phones with an active call hear the call-waiting tone.

After a phone answers an incoming call, the answering phone is in the connected state. Other phones that share the octo-line directory number are in the remote-in-use state.

After a connected call on an octo-line directory number is put on-hold, any phone that shares this directory number can pick up the held call. If a phone user is in the process of initiating a call transfer or creating a conference, the call is locked and other phones that share the octo-line directory number cannot steal the call.

Octo-Line Directory Number shows an octo-line directory number for SCCP phones in Cisco Unified CME.

Figure 3. Octo-Line Directory Number

The Barge and Privacy features control whether other phones are allowed to view call information or join calls on the shared octo-line directory number.

Feature Comparison by Directory Number Line-Mode on SCCP Phones

Table 1 lists some common directory number features and their support based on the type of line mode defined with the ephone-dn command.

Table 1. Feature Comparison by Line Mode on SCCP Phones

Feature

Single-Line

Dual-Line

Octo-Line

Barge

—

—

Yes

Busy Trigger

—

—

Yes

Conferencing (8-party)

—

4 directory numbers

1 directory number

FXO Trunk Optimization

Yes

Yes

—

Huntstop Channel

—

Yes

Yes

Intercom

Yes

—

—

Key System (one call per button)

Yes

—

—

Maximum Calls

—

—

Yes

MWI

Yes

—

—

Overlay directory numbers (c, o, x)

Yes

Yes

—

Paging

Yes

—

—

Park

Yes

—

—

Privacy

—

—

Yes

SIP Shared-Line (Nonexclusive)

Cisco Unified CME 7.1 and later versions support SIP shared lines to allow multiple phones to share a common directory number. All phones sharing the directory number can initiate and receive calls at the same time. Calls to the shared line ring simultaneously on all phones without active calls and any of these phones can answer the incoming calls. After a phone answers a call, the ringing stops on all phones and the call-waiting tone plays for other incoming calls to the connected phone.

The phone that answers an incoming call is in the connected state. Other phones that share the directory number are in the remote-in-use state. The first user that answers the call on the shared line is connected to the caller and the remaining users see the call information and status of the shared line.

Calls on a shared line can be put on hold like calls on a non-shared line. When a call is placed on hold, other phones with the shared-line directory number receive a hold notification so all phones sharing the line are aware of the held call. Any shared-line phone user can resume the held call. If the call is placed on hold as part of a conference or call transfer operation, the call cannot be resumed by other shared-line phone users. The ID of the held call is used by other shared-line members to resume the call. Notifications are sent to all associated phones when a held call is resumed on a shared line.

Shared lines support up to 16 calls, depending on the configuration in Cisco Unified CME, which rejects any new call that exceeds the configured limit. For configuration information, see Create Directory Numbers for SIP Phones.

The Barge and Privacy features control whether other phones are allowed to view call information or join calls on the shared-line directory number. See Barge and Privacy.


Note

When the no supplementary-service sip handle-replaces command is configured, SIP shared-line is not supported on CME.

Two Directory Numbers with One Telephone Number

Two directory numbers with one telephone or extension number have the following characteristics:

  • Have the same telephone number but two separate virtual voice ports, and therefore can have two separate call connections.

  • Can be dual-line (SCCP only) or single-line directory numbers.

  • Can appear on the same phone on different buttons or on different phones.

  • Should be used when you want the ability to make more call connections while using fewer numbers.

Two Directory Numbers with One Number on One Phone shows a phone with two buttons that have the same number, extension 1003. Each button has a different directory number (button 1 is directory number 13 and button 2 is directory number 14), so each button can make one independent call connection if the directory numbers are single-line and two call connections (for a total of four) if the directory numbers are dual-line.

Two Directory Numbers with One Number on Two Phones shows two phones that each have a button with the same number. Because the buttons have different directory numbers, the calls that are connected on these buttons are independent of one another. The phone user at phone 4 can make a call on extension 1003, and the phone user on phone 5 can receive a different call on extension 1003 at the same time.

The two directory numbers-with-one-number situation is different than a shared line, which also has two buttons with one number but has only one directory number for both of them. A shared directory number will have the same call connection at all the buttons on which the shared directory number appears. If a call on a shared directory number is answered on one phone and then placed on hold, the call can be retrieved from the second phone on which the shared directory number appears. But when there are two directory numbers with one number, a call connection appears only on the phone and button at which the call is made or received. In the example in Two Directory Numbers with One Number on Two Phones, if the user at phone 4 makes a call on button 1 and puts it on hold, the call can be retrieved only from phone 4. For more information about shared lines, see Shared Line (Exclusive) section.

The examples in Two Directory Numbers with One Number on One Phone and Two Directory Numbers with One Number on Two Phones show how two directory numbers with one number are used to provide a small hunt group capability. In Two Directory Numbers with One Number on One Phone, if the directory number on button 1 is busy or does not answer, an incoming call to extension 1003 rolls over to the directory number associated with button 2 because the appropriate related commands are configured. Similarly, if button 1 on phone 4 is busy, an incoming call to 1003 rolls over to button 1 on phone 5.

Figure 4. Two Directory Numbers with One Number on One Phone
Figure 5. Two Directory Numbers with One Number on Two Phones

Dual-Number

A dual-number directory number has the following characteristics:

  • Has two telephone numbers, a primary number and a secondary number.

  • Can make one call connection if it is a single-line directory number.

  • Can make two call connections at a time if it is a dual-line directory number (SCCP only).

  • Should be used when you want to have two different numbers for the same button without using more than one directory number.

Dual-Number Directory shows a directory number that has two numbers, extension 1006 and extension 1007.

Figure 6. Dual-Number Directory

Shared Line (Exclusive)

An exclusively shared directory number has the following characteristics:

  • Has a line that appears on two different phones but uses the same directory number, and extension or phone number.

  • Can make one call at a time and that call appears on both phones.

  • Should be used when you want the capability to answer or pick up a call at more than one phone.

Because this directory number is shared exclusively among phones, if the directory number is connected to a call on one phone, that directory number is unavailable for calls on any other phone. If a call is placed on hold on one phone, it can be retrieved on the second phone. This is like having a single-line phone in your house with multiple extensions. You can answer the call from any phone on which the number appears, and you can pick it up from hold on any phone on which the number appears.


Note
Transcoding is not supported for Shared Lines. From Unified CME Release 12.2, you can use Voice Class Codec (VCC) with shared lines.

Shared Directory Number (Exclusive) shows a shared directory number on phones that are running SCCP. Extension 1008 appears on both phone 7 and phone 8.

Figure 7. Shared Directory Number (Exclusive)

Shared Lines with Voice Class Codec Support

From Unified CME 12.2 Release, Unified CME supports voice class codecs (VCC) with SIP shared lines. A VCC is a construct within which a codec preference order is defined. Preferences defined within the VCC can be used to determine which codecs will be selected over others. When a VCC is applied to a dial peer on the Unified CME, the dial peer then follows the preference order defined in the VCC.

The VCC configuration can be applied for phones having shared line configured on Unified CME. However, the voice class codec behavior of SIP trunk remains unchanged. It is recommended that the same voice class codec configuration is applied on all phones using the shared line directory number. The VCC configuration applied under the voice register pool configuration mode is used for filtering the codecs on inbound and outbound calls from the phone. If the VCC configuration does not have a common codec negotiated, then the call is disconnected. When the codec on the incoming SIP trunk is not listed in the VCC, the call is not placed. It is mandatory to configure the CLI command supplementary-service media-renegotiate under voice service voip configuration mode for VCC configuration support with SIP shared lines. For a sample configuration of VCC with shared line, see Examples for Configuring VCC with Shared Lines.

Codec Support

All the codecs listed under the CLI command voice class codec are supported as part of the VCC support for SIP shared lines on Unified CME.

Feature Support
The following shared line features are supported as part of the VCC configuration:

  • Hold and Remote Resume

  • Barge

  • cBarge

  • Video

  • MOH Transcoding

  • Privacy

Advantages

  • Insertion of transcoding resource to place a call can be avoided.

Restrictions

  • Transcoding is not supported for SIP shared lines with VCC support.

Mixed Shared Lines

Cisco Unified CME 9.0 and later versions support the mixed Cisco Unified SIP/SCCP shared line. This feature allows Cisco Unified SIP and SCCP IP phones to share a common directory number.

The mixed shared line supports up to 16 calls, depending on the configuration in Cisco Unified CME, which rejects any new call that exceeds the configured limit.

For configuration information, see Create Directory Numbers for SCCP Phones and Create Directory Numbers for SIP Phones.

Incoming and Outgoing Calls

All phones sharing the common directory number can initiate and receive calls at the same time. Calls to the mixed shared line ring simultaneously on all phones without active calls and any of these phones can answer the incoming calls. After a phone answers a call, the ringing stops on all phones and the call-waiting tone plays for other incoming calls to the connected phone.

The phone that answers an incoming call is in the connected state. Other phones that share the common directory number are in the remote-in-use state. The first user who answers the call on the mixed shared line is connected to the caller and the remaining users see the call information and status of the mixed shared line.

When a mixed shared-line user makes an outgoing call on the shared line, all the other shared-line users are notified of the outgoing call. When the called party answers, the caller is connected while the remaining shared-line users see the call information and the status of the call on the mixed shared line.

Hold and Resume

Calls on a mixed shared line can be put on hold like calls on a non-shared line. When a call is placed on hold, other phones with the shared-line directory number receive a hold notification so all phones sharing the line are aware of the call on hold. Any shared-line phone user can resume the call on hold. The ID of the call on hold is used by other shared-line members to resume the call. Notifications are sent to all associated phones when a call on hold is resumed on a mixed shared line. If the call is placed on hold as part of a conference or call transfer operation, the resume feature is not allowed.

Privacy on Hold

The Privacy on Hold feature prevents other phone users from viewing call information or retrieving a call put on hold by another phone sharing a common directory number. Only the caller who put the call on hold can see the status of the held call.

By default, Privacy on Hold feature is disabled for all phones on a shared line. Use the privacy-on-hold command in telephony-service configuration mode to enable the Privacy feature for calls that are on hold on Cisco Unified SCCP IP phones on a mixed shared line. Use the privacy-on-hold command in voice register global configuration mode to enable the Privacy feature for calls that are on hold on Cisco Unified SIP IP phones on a mixed shared line.

The no privacy and privacy off commands override the privacy-on-hold command.

Call Transfer and Forwarding

Both blind transfer and consult transfer are supported on a mixed shared line. A mixed shared line can be the one transferring the call, the one receiving the transferred call, or the call being transferred.

There are four types of call forwarding: all calls, no answer, busy, and night service. Any of these can be configured under a shared SCCP ephone-dn or a shared SIP voice register dn. However, the user must keep the call forwarding parameters for the SCCP and SIP lines synchronized with each other. A mixed shared line can be the one forwarding the call, the one receiving the forwarded call, or the call being forwarded.

For more information, see Configure Call Transfer and Forwarding.

Call Pickup

The Call Pickup feature is supported on a mixed shared line and SIP lines only when the call-park system application command is configured in telephony-service configuration mode.

A user can answer a call that:

  • Originates from a shared line

  • Rings on a shared line

  • Originates from one shared line and rings on another shared line

For more information, see Call Pickup.

Call Park

The Call Park feature is supported on a mixed shared line and SIP lines only when the call-park system application command is configured in telephony-service configuration mode.

For more information, see Call Park.

Message Waiting Indication

SCCP and SIP message-waiting indication (MWI) services are supported on Cisco Unity and Cisco Unity voice mails on mixed shared lines:

The following are two ways of registering a mixed shared line for an MWI service from a SIP-based MWI server with the shared-line option:

  • Configure the mwi sip command in ephone-dn or ephone-dn-template configuration mode.

  • Configure the mwi command in voice register dn configuration mode.

For SCCP MWI service on a mixed shared line, use the mwi { off | on | on-off} command in ephone-dn configuration mode to enable a specific Cisco Unified IP phone extension to receive MWI notification from an external voice-messaging system.

Software Conferencing

A local software conference can be created on a mixed shared line, with the mixed shared line acting as a conference creator and a conference participant.

For software conferencing on a mixed shared line, other shared-line users remain in remote-in-use state and do not see the calls on hold when the conference call is put on hold by a mixed-shared-line user acting as the conference creator.


Note

Only the conference creator, who put a conference call on hold, can resume the conference call.

Dial Plan

A dial plan pattern enables abbreviated extensions to be expanded into fully qualified E.164 numbers and builds additional dial peers for the expanded numbers it creates.

Features are effectively supported on a mixed shared line when dial-plan patterns have matching configurations in telephony-service and voice register global configuration modes using the dialplan pattern command.

Busy Lamp Field Speed-Dial Monitoring

A mixed shared line only supports directory number-based Busy-Lamp-Field (BLF) Speed-Dial monitoring and not device-based monitoring.

Restrictions For Mixed Shared Lines

The following features are not supported on mixed Cisco Unified SIP/SCCP shared lines:

  • Single Number Reach

  • Hardware Conferencing

  • Remote-resume on a local software conference call

  • Video calls

  • Overlay DNs on Cisco Unified SCCP IP phones

Feature Support

The following features are supported on mixed Cisco Unified SIP/SCCP shared lines from Unified CME, 12.2:

  • Hold and Resume

  • Privacy

  • Barge

  • cBarge

Overlaid Directory Numbers

An overlaid directory number has the following characteristics:

  • Is a member of an overlay set, which includes all the directory numbers that have been assigned together to a particular phone button.

  • Can have the same telephone or extension number as other members of the overlay set or different numbers.

  • Can be single-line or dual-line, but cannot be mixed single-line and dual-line in the same overlay set.

  • Can be shared on more than one phone.

Overlaid directory numbers provide call coverage similar to shared directory numbers because the same number can appear on more than one phone. The advantage of using two directory numbers in an overlay arrangement rather than as a simple shared line is that a call to the number on one phone does not block the use of the same number on the other phone, as would happen if it were a shared directory number.

For information about configuring call coverage using overlaid ephone-dns, see Configure Call Coverage Features.

You can overlay up to 25 lines on a single button. A typical use of overlaid directory numbers would be to create a “10x10” shared line, with 10 lines in an overlay set shared by 10 phones, resulting in the possibility of 10 simultaneous calls to the same number. For configuration information, see Creating Directory Numbers for a Simple Key System on SCCP Phone.

Auto Registration of SIP Phones on Cisco Unified CME

Cisco Unified CME supports auto registration of both SIP and SCCP phones. When the auto registration feature is enabled, the voice register pool and voice register dn commands do not need to be manually configured for the phones. The configuration is automatically created when the phone registers.

The auto registration feature for SIP phones is enabled with the auto-register command under voice register global configuration mode. For more information on auto-register command, see Cisco Unified Communications Manager Express Command Reference.

The auto registration of SCCP phones is enabled with the auto-reg-ephone command under telephony-service configuration mode. For more information on auto-register command, see Cisco Unified Communications Manager Express Command Reference.

As part of the auto-register command, certain CLI sub-mode configuration options are available to the administrator to successfully register phones using auto-registration on Unified CME. 


Router(config-register-global)#auto-register
Router(config-voice-auto-register)#
Router(config-voice-auto-register)# ?
VOICE auto register configuration commands:
  auto-assign     Define DN range for auto assignment
  default         Set a command to its defaults
  exit            Exit from voice register group configuration mode
  no              Negate a command or set its defaults
  password        Default password for auto-register phones
  service-enable  Enable SIP phone Auto-Registration
  template        Default template for auto-register phones

For details on the configuration steps for auto registration of SIP phones, see Configure Auto Registration for SIP Phones.

Service Enable —If the administrator needs to temporarily disable or enable auto registration without losing configurations such as DN range, and password, the no form of the CLI option service-enable is used (no service-enable ). Once auto-register command is entered, the service is enabled by default. To re-enable the auto registration feature, use the command service-enable . It is a sub-mode option in the CLI command auto-register . To disable auto registration including removal of configurations such as password and DN range, the no form of the CLI command auto-register (under voice register global) is used. 


Router(config)#voice register global
Router(config-register-global)#auto-register
Router(config-voice-auto-register)#no service-enable ?
<cr>

Password —As part of the auto registration feature, authentication of phones registering on Unified CME is enabled. When the phone registers with Unified CME, it is mandatory for the administrator to configure the password credentials; username is assigned by default. However, the administrator can modify the username and password credentials under the corresponding voice register pool that gets created after auto registration. 


Router(config)#voice register global
Router(config-register-global)#auto-register
Router(config-voice-auto-register)#password ?
  WORD  Password string

Note

It is mandatory that password is configured before DN range (auto-assign) while registering phones using auto registration.

Auto Assign —It is mandatory to define a directory number (DN) range for auto-registration feature to work. The DN range that can be assigned to phones registering on Unified CME is configured using auto-assign <first-dn > to <last-dn >, which is a submode option of the CLI command auto-register (under voice register global ). The DN numbers assigned to the phones through auto registration are always within the DN range that is defined. However, ensure that the defined DN range is within the maximum DNs recommended for the supported platform. 


Router(config)#voice register global
Router(config-register-global)#auto-register
Router(config-voice-auto-register)#auto-assign ?
  <1-4294967295>  First DN number
Router(config-voice-auto-register)#auto-assign 1001 ?
  <1-4294967295>  Last DN number
Router(config-voice-auto-register)#auto-assign 1001 to 1010

The automatic registration feature also provides the administrators with the option to enhance a predefined DN range. The enhancement of an existing DN range is supported such that the new first-dn is not greater that the existing first-dn and the new last-dn is not less than the existing last-dn.

For example, the DN range 8001-8006 can be enhanced as 7999-8006, 8000-8007, but not as 8002-8006 or 8001 to 8005. 

Router# show running-config | section voice register global 
	 voice register global 
	  mode cme 
	  source-address 8.41.20.1 port 5060 
	 		 auto-register 
	 		 		 password xxxx 
	 		 	 auto-assign 8001 to 8006 
	  max-dn 50 
	  max-pool 40
	 Router(config-register-global)#auto-assign 8002 to 8006
	 Start DN should not be greater than existing First DN 
	 Router(config-register-global)#auto-assign 8001 to 8005  
	 Stop DN should not be less than existing Last DN

The DN assigned to phone using the auto registration feature does not duplicate a manually configured DN. When the defined DN range includes a previously registered DN, that DN is skipped as part of the auto registration process. However, when a previously registered DN deregisters and the corresponding configuration for the DN and pool are removed, it can be assigned to a phone registering on Unified CME using auto registration. The assignment of DN range is done in round robin fashion and the first available free DN is assigned to the phone that is auto registering with Unified CME.


Note

We recommend that administrators choose different DN ranges for manually configured and auto configured phones.

Template —Administrators are provided the option to create a basic configuration template that can be applied to all phones registering automatically on Unified CME. This basic configuration template supports all the configurations currently supported by the voice register template. It is mandatory that voice register template is configured with the same template tag. 


Router(config)#voice register global
Router(config-register-global)#auto-register
Router(config-voice-auto-register)#template ?
  <1-10>  template tag>
Router(config-voice-auto-register)#template 10

All phone configurations such as voice-register-pool and voice-register-dn that are generated as part of the auto registration process are persistent configurations. These configurations will be available on the Unified CME even after an event of router reload.

The CLI commands show voice register pool all and show voice register pool all brief distinctly mention the registration process for phones as registered or unregistered for manual registration, and registered* or unregistered* for automatic registration. However, the registration status for auto-registered phones are reset in the event of a router reload. Then, phone registration status displays only as registered or unregistered.

Syslog Messages

Unified CME generates Syslog messages as part of the registration feature, when the phone registers and unregisters with the Cisco Unified CME. Also, based on the DN range configured, the administrator gets syslog message providing updates on the registration status of assigned DNs. The syslog messages that provide updates are generated at two instances; at 80% utilization of available DNs, and at 100% utilization of DNs.

From Unified CME 12.3 Release (Cisco IOS XE Fuji Release 16.9.1), the following changes are introduced to Syslog messages printed in Unified CME:

  • Syslog messages are printed for successful endpoint assignment and unassignment using Extension Assigner (EA) feature.

  • The device type information in the registration and unregistration syslog messages of Unified CME is printed as DeviceType:Phone-Type

A sample output for Unified CME 12.3 syslog changes is as follows:


Successful extension assignment:
=====================
000246: *Apr 23 03:58:46.238: %EXTASSIGNER-6-ASSIGNED: Extension assignment successful for phone:SEP382056447710. New pool(2). Old pool(1).     

Successful extension un-assignment:
=======================
000407: *May  3 07:13:08.876: %EXTASSIGNER-6-UNASSIGNED: Extension unassignment successful for phone:SEP382056447710. Unassigned pool(2).                                    
                                

Phone un-registration:
==============
000300: *Apr 23 03:58:55.128: %SIPPHONE-6-UNREGISTER: VOICE REGISTER POOL-1 has unregistered. Name:SEP382056447710  IP:8.55.0.108  DeviceType:Phone-8851

Phone registration:
============
000310: *Apr 23 03:59:08.054: %SIPPHONE-6-REGISTER: VOICE REGISTER POOL-2 has registered. Name:SEP382056447710  IP:8.55.0.108  DeviceType:Phone-8851

The Unified CME system generates the following syslog messages as part of auto registration.

  • Syslog message when phone registers with Unified CME: 

    *Mar 28 21:44:08.795 IST: %SIPPHONE-6-REGISTER: VOICE REGISTER POOL-8 has registered. Name:SEP2834A2823843  IP:8.41.20.58 DeviceType:Phone

  • Syslog message at 80% utilization of DN range: 

    *Mar 28 21:42:25.732 IST: %SIPPHONE-6-AUTOREGISTER80: AUTO-REGISTER: 80% of DN range is consumed

  • Syslog message at 100% utilization of DN range: 

    *Mar 28 21:44:03.328 IST: %SIPPHONE-6-AUTOREGISTER100: AUTO-REGISTER: 100% of DN range is consumed

  • Syslog message when phone unregisters with Unified CME: 

    *Mar 28 18:03:41.748 IST: %SIPPHONE-6-UNREGISTER: VOICE REGISTER POOL-6 has unregistered. Name:SEPB000B4BAF3DA  IP:8.41.20.53  DeviceType:Phone

Monitor Mode for Shared Lines

In Cisco CME 3.0 and later versions, monitor mode for shared lines provides a visible line status indicating whether the line is in-use or not. A monitor-line lamp is off or unlit only when its line is in the idle call state. The idle state occurs before a call is made and after a call is completed. For all other call states, the monitor line lamp is lit. A receptionist who monitors the line can see that it is in use and can decide not to send additional calls to that extension, assuming that other transfer and forwarding options are available, or to report the information to the caller; for example, “Sorry, that extension is busy, can I take a message?”

In Cisco CME 3.2 and later versions, consultative transfers can occur during Direct Station Select (DSS) for transferring calls to idle monitored lines. The receptionist who transfers a call from a normal line can press the Transfer button and then press the line button of the monitored line, causing the call to be transferred to the phone number of the monitored line. For information about consultative transfer with DSS, see Configure Call Transfer and Forwarding.

In Cisco Unified CME 4.0(1) and later versions, the line button for a monitored line can be used as a DSS for a call transfer when the monitored line is idle or in-use, provided that the call transfer can succeed; for example, when the monitored line is configured for Call Forward Busy or Call Forward No Answer.


Note

Typically, Cisco Unified CME does not attempt a transfer that causes the caller (transferee) to hear a busy tone. However, the system does not check the state of subsequent target numbers in the call-forward path when the transferred call is transferred more than once. Multiple transfers can occur because a call-forward-busy target is also busy and configured for Call Forward Busy.

In Cisco Unified CME 4.3 and later versions, a receptionist can use the Transfer to Voicemail feature to transfer a caller directly to a voice-mail extension for a monitored line. For configuration information, see Transfer to Voice Mail.

For configuration information for monitor mode, see Create Directory Numbers for SCCP Phones.

Monitor mode is intended for use only in the context of shared lines so that a receptionist can visually monitor the in-use status of several users’ phone extensions; for example, for Busy Lamp Field (BLF) notification. To monitor all lines on an individual phone so that a receptionist can visually monitor the in-use status of that phone, see Watch Mode for Phones.

For BLF monitoring of speed-dial buttons and directory call-lists, see Configure Presence Service.

Watch Mode for Phones

In Cisco Unified CME 4.1 and later versions, a line button that is configured for watch mode on one phone provides BLF notification for all lines on another phone (watched phone) for which watched directory number is the primary line. Watch mode allows a phone user, such as a receptionist, to visually monitor the in-use status of an individual phone. A user can use the line button that has been set in watch mode as a speed-dial to call the first extension of the watched phone. The watching phone button displays a red light when the watched phone is unregistered in a DND state or in an offhook state. Pressing the button when it is not displaying a red light will dial the number in the same manner it would for a monitor button or the speed-dial button. Incoming calls on a line button that is in watch mode do not ring and do not display caller ID or call-waiting caller ID.

The line button for a watched phone can also be used as a DSS for a call transfer when the watched phone is idle. In this case, the phone user who transfers a call from a normal line can press the Transfer button and then press the line button of the watched directory number, causing the call to be transferred to the phone number associated with the watched directory number.

For configuration information, see Create Directory Numbers for SCCP Phones.

If the watched directory number is a shared line and the shared line is not idle on any phone with which it is associated, then in the context of watch mode, the status of the line button indicates that the watched phone is in use.

For best results when monitoring the status of an individual phone based on a watched directory number, the directory number configured for watch mode should not be a shared line. To monitor a shared line so that a receptionist can visually monitor the in-use status of several users’ phone extensions, see Monitor Mode for Shared Lines.

For BLF monitoring of speed-dial buttons and directory call-lists, see Presence Service.

PSTN FXO Trunk Lines

In Cisco CME 3.2 and later versions, IP phones running SCCP can be configured to have buttons for dedicated PSTN FXO trunk lines, also known as FXO lines. FXO lines may be used by companies whose employees require private PSTN numbers. For example, a salesperson may need a special number that customers can call without having to go through a main number. When a call comes in to the direct number, the salesperson knows that the caller is a customer. In the salesperson’s absence, the customer can leave a voice mail. FXO lines can use PSTN service provider voice mail: when the line button is pressed, the line is seized, allowing the user to hear the stutter dial tone provided by the PSTN to indicate that voice messages are available.

Because FXO lines behave as private lines, users do not have to dial a prefix, such as 9 or 8, to reach an outside line. To reach phone users within the company, FXO-line users must dial numbers that use the company's PSTN number. For calls to non-PSTN destinations, such as local IP phones, a second directory number must be provisioned.

Calls placed to or received on an FXO line have restricted Cisco Unified CME services and cannot be transferred by Cisco Unified CME. However, phone users are able to access hookflash-controlled PSTN services using the Flash softkey.

In Cisco Unified CME 4.0(1), the following FXO trunk enhancements were introduced to improve the keyswitch emulation behavior of PSTN lines on phones running SCCP in a Cisco Unified CME system:

  • FXO port monitoring—Allows the line button on IP phones to reliably show the status of an FXO port when the port is in use. The status indicator, either a lamp or an icon, depending on the phone model, accurately displays the status of the FXO port during the duration of the call, even after the call is forwarded or transferred. The same FXO port can be monitored by multiple phones using multiple trunk ephone-dns.

  • Transfer recall—If a transfer-to phone does not answer after a specified timeout, the call is returned to the phone that initiated the transfer and it resumes ringing on the FXO line button. The directory number must be dual-lined.

  • Transfer-to button optimization—When an FXO call is transferred to a private extension button on another phone, and that phone has a shared line button for the FXO port, after the transfer is committed and the call is answered, the connected call displays on the FXO line button of the transfer-to phone. This frees up the private extension line on the transfer-to phone. The directory number n must be dual-line.

  • Dual-line ephone-dns— Directory numbers for FXO lines can now be configured for dual-line to support the FXO monitoring, transfer recall, and transfer-to button optimization features.

For configuration information, see Configure Trunk Lines for a Key System on SCCP Phone.

Codecs for Cisco Unified CME Phones

In Cisco CME 3.4, support for connecting and provisioning SIP phones was added. The default codec of the POTS dial peer for an SCCP phone is G.711 and the default codec of a VoIP dial peer for a SIP phone is G.729. If neither the SCCP phone nor the SIP phone in Cisco Unified CME is specifically configured to change the codec, calls between the two phones on the same router will produce a busy signal caused by the mismatched default codecs. To avoid codec mismatch, specify the codec for individual IP phones in Cisco Unified CME. Modify the configuration for either SIP or SCCP phones to ensure that the codec for all phones match. Do not modify the configuration for both SIP and SCCP phones. For configuration information, see Configure Codecs of Individual Phones for Calls Between Local Phones.

In Cisco Unified CME 4.3, support for G.722-64K and the Internet Low Bit Rate Codec (iLBC) was added. This enables Cisco Unified CME to support the same codecs that are used in newer Cisco Unified IP phones, mobile wireless networks, and internet telephony without transcoding. This feature provides support for the following:

  • iLBC and G.722-capable SIP and SCCP IP phones in Cisco Unified CME.

  • iLBC-capable SCCP analog endpoints and remote phones in Cisco Unified CME.

  • Conferencing support for G.722 and ILBC.

  • Supplementary services, such as transfer, call forward, MOH, support for G.722 and iLBC, including any supplementary services that require transcoding between G.722 and any other codec.

  • Transcoding for G.722 and iLBC, including G.722 to G.711 and G.722 to any other codec.

With the introduction of G.722 and iLBC codecs, there can be a disparity between codec capabilities of different phones and different firmware versions on same phone type. For example, when a H.323 call is established, the codec is negotiated based on the dial-peer codec and the assumption is that the codecs supported on H.323 side are supported by the phones. This assumption is not valid after G.722 and ILBC codec are introduced in your network. If the phones do not support the codecs on the H.323 side, a transcoder is required. To avoid transcoding in this situation, configure incoming dial-peers so that G.722 and iLBC codecs are not used for calls to phones that are not capable of supporting these codecs. Instead, configure these phones for G.729 or G.711. Also, when configuring shared directory numbers, ensure that phones with the same codec capabilities are connected to the shared directory number.

G.722-64K

Traditional PSTN telephony codecs, including G.711 and G.729, are classified as narrowband codecs because they encode audio signals in a narrow audio bandwidth, giving telephone calls a characteristic “tinny” sound. Wideband codecs, such as G.722, provide a superior voice experience because wideband frequency response is 200 Hz to 7 kHz compared to narrowband frequency response of 300 Hz to 3.4 kHz. At 64 kbps, the G.722 codec offers conferencing performance and good music quality.

A wideband handset for certain Cisco Unified IP phones, such as the Cisco Unified IP Phone 7906G, 7911G, 7941G-GE, 7942G, 7945G, 7961G-GE, 7962G, 7965G, and 7975G, take advantage of the higher voice quality provided by wideband codecs to enhance end-user experience with high-fidelity wideband audio. When users use a headset that supports wideband, they experience improved audio sensitivity when the wideband setting on their phones is enabled. You can configure phone-user access to the wideband headset setting on IP phones by setting the appropriate VendorConfig parameters in the phone’s configuration file. For configuration information, see Modify Cisco Unified IP Phone Options.

If the system is not configured for a wideband codec, phone users may not detect any additional audio sensitivity, even when they are using a wideband headset.

You can configure the G.722-64K codec at a system-level for all calls through Cisco Unified CME. For configuration information, see Modify the Global Codec. To configure individual phones and avoid codec mismatch for calls between local phones, see Configure Codecs of Individual Phones for Calls Between Local Phones.

iLBC codec

Internet Low Bit Rate Codec (iLBC) enables graceful speech quality degradation in a network where frames get lost. Consider iLBC suitable for real-time communications, such as telephony and video conferencing, streaming audio, archival, and messaging. This codec is widely used by internet telephony softphones.The SIP, SCCP, and MGCP call protocols support use of the iLBC as an audio codec. iLBC provides better voice quality than G.729 but less than G.711. Supporting codecs that have standardized use in other networks, such as iLBC, enables end-to-end IP calls without the need for transcoding.

To configure individual SIP or SCCP phones, including analog endpoints in Cisco Unified CME, and avoid codec mismatch for calls between local phones, see Configure Codecs of Individual Phones for Calls Between Local Phones.

Analog Phones

Cisco Unified CME supports analog phones and fax machines using Cisco Analog Telephone Adaptors (ATAs) or FXS ports in SCCP, H.323 mode, and fax pass-through mode. The FXS ports used for analog phones or fax can be on a Cisco Unified CME router, Cisco VG224 voice gateway, or integrated services router (ISR).

This section provides information on the following topics:

Cisco ATAs in SCCP Mode

You can configure the Cisco ATA 186 or Cisco ATA 188 to cost-effectively support analog phones using SCCP in Cisco IOS Release 12.2(11)T and later versions. Each Cisco ATA enables two analog phones to function as IP phones. For configuration information, see Configure Cisco ATA Support in SCCP Mode.

Cisco ATAs in SIP Mode

You can configure the Cisco ATA 187, Cisco ATA 190 or Cisco ATA 191 to cost-effectively support analog phones and FAX using SIP for Unified CME. The support for Cisco ATA 191 is introduced from Unified CME 12.5 (Cisco IOS XE Gibralter 16.10.1a) Release. Each Cisco ATA enables two analog phones to function as IP phones. For configuration information, see Configure Cisco ATA Support in SIP Mode.

The following are some of the known restrictions for Cisco ATA 191 on Unified CME:

  • If both ports of a Cisco ATA 191 are configured as shared line, then a call put on hold on one port cannot be resumed at the other port.

  • For Unified CME, a call put on hold on Unified SIP IP Phone cannot be resumed from a Cisco ATA 191.

  • You cannot configure the same shared line DN on both ports of the Cisco ATA 191. On configuring the same shared line DN on both the lines of the Cisco ATA 191, second line does not get registered.

Cisco ATA 191 on Unified CME

The ATA 191 analog telephone adapter is a telephony-device-to-Ethernet adapter that allows regular analog phones to operate on IP-based telephony networks. The ATA 191 supports two voice ports, each with an independent phone number. The ATA 191 also has an RJ-45 10/100BASE-T data port.

Unified CME 12.5 and later release provide native support for Cisco ATA 191. The SIP protocol is supported on Cisco ATA 191.

The ATA 191 supports two lines, but has only a single MAC address. Hence, you must use a shifted MAC address to configure the second line on ATA 191. A sample configuration for Line 1 and Line 2 for an ATA 191 is as follows: 


Line 1 confguration:
voice register dn 15
    number 8015
voice register pool 15
    id mac DCEB.941C.F33D
    type ATA-191
    number 1 dn 15
    username abcd password xxxx
    codec g711ulaw 
Line 2 confguration:
voice register dn 16
    number 8016
    voice register pool 16
    id mac EB94.1CF3.3D01
    type ATA-191
    number 1 dn 16
    username uvwx password xxxx
    codec g711ulaw

Note

Left shift the MAC address by two places, and append the two removed digits at the end with 01 to define the shifted MAC address. For example, the MAC address DCEB.941C.F33D is modified to get the shifted MAC address, EB94.1CF3.3D01.

Feature Support for Cisco ATA 191

The Cisco ATA 191 supports the following features on Unified CME:

  • Hold or Resume—Hold or Resume is invoked using a hookflash for Cisco ATA 191 on Unified CME. For more information on the feature, see Put a Call on Hold on Your Analog Phone.

  • Consult or Semi Consult Transfer—To Transfer a call using Cisco ATAT 191 on Unified CME, you need to use hookflash along with FAC. For information on the feature, see Transfer a Call from Your Analog Phone.

  • Call Waiting—Call Waiting calls are answered using a hookflash for Cisco ATA 191 on Unified CME. For more information on the feature, see Answer Call Waiting on Your Analog Phone.

  • MeetMe Conference—To host a MeetMe Conference on Cisco ATAT 191 on Unified CME, you need to use hookflash along with FAC. For information on how to invoke the feature, see Host a Meet Me Conference on Your Analog Phone.

  • Call Forward (All, Busy, No Answer)—Call Forward is invoked using a hookflash for Cisco ATA 191 on Unified CME. For more information on the feature, see Forward Your Analog Phone Calls to Another Number.

  • cBarge—cBarge is invoked using a hookflash for Cisco ATA 191 on Unified CME. For more information on the feature, see Call Features and Star Codes for Analog Phones.

  • Built-in Bridge Conference (BIB)—BIB is invoked using a hookflash for Cisco ATA 191 on Unified CME. For more information on the feature, see Make a Conference Call from Your Analog Phone.

  • Call Park—Call Park is invoked using a FAC Code for Cisco ATA 191 on Unified CME. To park a call on Cisco ATA 191 on Unified CME, you need to transfer the call to the FAC code, **6. For more information, see Call Park.

  • Call Park Pickup and G-Pickup—To pick up a parked call, dial the park-slot number.

  • Voice Mail—For Voice Mail support on Cisco ATA 191, you need to go offhook, and dial the voice mail number configured on Unified CME to access the IVR options.

  • Fax Transmission (with T.38, Passthrough)—For Fax trasmission to work with Cisco ATA 191 on Unified CME, you need to configure the CLI command service phone faxMode 0 under telephony-service configuration mode. For information on the feature, see Send and Receive Fax Calls.

  • Shared Line/Mixed Shared Line—For information on the feature, see Shared Lines on Your Analog Phone.

  • KPML Dialing—For KPML Dialing support on Cisco ATA 191, you need to go offhook and dial the number.

  • TCP/UDP Registration

  • Extension Assigner

  • Auto Registration

  • DTMF

  • Caller ID Blocking

  • Music On Hold (MOH)

  • Upgrade or Downgrade Firmware

  • Redial

  • WebAccess

  • SSH

  • MWI—Cisco ATA 191 plays a stuttered tone instead of MWI

Feature Support Restriction

The following are the known feature restrictions for Cisco ATA 191 on Unified CME:

  • Barge—Cisco ATA 191 cannot barge into an active shared line call (phone limitation). However, non-ATA phones can barge into Cisco ATA's shared line call.

  • Hardware Conference is not supported.

  • Do Not Disturb

  • Span to PC Port

  • Speed Dial—For Cisco ATA 191, Abbreviated Dial is supported as Speed Dial. Unified CME does not support Abbreviated Dial.

  • Secondary CME

  • Call Waiting with Caller-ID—For Cisco ATA 191, the phone Caller-ID does not display any call waiting notification (only call waiting tone is supported).

  • Localization

  • Shared Line

    • Both the ports of a Cisco ATA191 cannot be configured with the same Shared Line DN.

    • Remote Resume is not supported for a Shared Line call placed on hold.

FXS Ports in H.323 Mode

FXS ports on platforms that cannot enable SCCP supplementary features can use H.323 mode to support call waiting, caller ID, hookflash transfer, modem pass-through, fax (T.38, Cisco fax relay, and pass-through), and PLAR. These features are provisioned as Cisco IOS voice features and not as Cisco Unified CME features.


Note

When using Cisco Unified CME, you can configure FXS ports in H.323 mode for call waiting or hookflash transfer, but not both at the same time.

Fax Support

Cisco Unified CME 4.0 introduced the use of G.711 fax pass-through for SCCP on the Cisco VG224 voice gateway and Cisco ATA. In Cisco Unified CME 4.0(3) and later versions, fax relay using the Cisco-proprietary fax protocol is the only supported fax option for SCCP-controlled FXS ports on the Cisco VG224 and integrated service routers. For more information on fax relay, see Fax Relay.

Cisco ATA-187

Cisco Unified CME 9.0 and later versions provide voice and fax support on Cisco ATA-187.

Cisco ATA-187 is a SIP-based analog telephone adaptor that turns traditional telephone devices into IP devices. Cisco ATA-187 can connect with a regular analog FXS phone or fax machine on one end, while the other end is an IP side that uses SIP for signaling and registers to Cisco Unified CME as a Cisco Unified SIP IP phone.

Cisco ATA-187 functions as a Cisco Unified SIP IP phone that supports T.38 fax relay and fax pass-through, enabling the real-time transmission of fax over IP networks. The fax rate is from 7.2 to 14.4 kbps.

For information on how to configure voice and fax support on Cisco ATA-187, see Configure Voice and T.38 Fax Relay on Cisco ATA-187.

For information on the features supported in Cisco ATA-187, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified Secure SRST.

For more information on Cisco ATA-187, see Cisco ATA 187 Analog Telephone Adaptor Administration Guide for SIP.

Cisco VG202, VG204, and VG224 Auto Configuration

The Auto Configuration feature in Cisco Unified CME 7.1 and later versions allows you to automatically configure the Cisco VG202, VG204, and VG224 Analog Phone Gateway. You can configure basic voice gateway information in Cisco Unified CME, which then generates XML configuration files for the gateway and saves the files to either the default location in system:/its/ or to a location you define in system memory, flash memory, or an external TFTP server. When the voice gateway powers up, it downloads the configuration files from Cisco Unified CME and based on the information in the files, the voice gateway provisions its analog voice ports and creates the corresponding dial peers.

Using this Auto Configuration feature with the existing Auto Assign feature allows you to quickly set up analog phones to make basic calls. After the voice gateway is properly configured and it downloads its XML configuration files from Cisco Unified CME, the SCCP telephony control (STC) application registers each configured voice port to Cisco Unified CME.

If you enable the Auto Assign feature, the gateway automatically assigns the next available directory number from the pool set by the auto assign command, binds that number to the requesting voice port, and creates an ephone entry associated with the voice port. The MAC address for the ephone entry is calculated based on the MAC address of the gateway and the port number. You can manually assign a directory number to each of the voice ports by creating the ephone-dn and corresponding ephone entry.

You can initiate a reset or restart of the analog endpoints from Cisco Unified CME, which triggers the autoconfiguration process. The voice gateway downloads its configuration files from Cisco Unified CME and applies the new changes.

For configuration information, see Auto-Configuration for Cisco VG202, VG204, and VG224.

Internet Protocol - Secure Telephone Equipment Support

Cisco Unified CME 8.0 adds support for a new secure endpoint, Internet Protocol - Secure Telephone Equipment (IP-STE). IP-STE is a standalone, V.150.1 capable device which functions like a 7960 phone with secure communication capability. IP-STE has native state signaling events (SSE / SPRT) support and supports SCCP protocol. IP-STE uses the device ID 30035 when registering to a SCCP server. However, only V.150.1 modem relay is implemented in an IP-STE stack and V150.1 modem passthrough is not supported. Therefore, the response to capability query from Cisco Unified CME only includes media_payload_XV150_MR_711U and media_payload_xv150_MR_729A.

For configuration information, see Configure Secure IP Phone (IP-STE) on SCCP Phone.

The following support is added for IP-STE endpoints:

  • The IP-STE endpoint allows secure communication between gateway-connected legacy analog STE/STU devices and IP STE devices using existing STE devices in voice networks.

  • Secure voice and secure data modes from STE/STU devices connected to Cisco IOS gateway foreign exchange station (FXS) and BRI ports to an IP-STE.

  • Support for the state signaling events (SSE) protocol, allowing for modem signaling end-to-end and VoIP to modem over IP (MoIP) transition and operation.

  • Interoperation between line-side and trunk-side gateways and Cisco Unified CME to determine codec support and V.150.1 negotiation. You can configure gateway-attached devices to support either modem relay, modem pass-through, both modem transport methods, or neither method.

Secure Communications Between STU, STE, and IP-STE

Secure Telephone Equipment (STE) and Secure Telephone Units (STUs) encrypt voice and data streams with government proprietary algorithms (Type-1 encryption). To provide support for the legacy STEs and STUs and next generation IP Secure Telephone Equipment (IP-STE), voice gateways must be able to support voice and data in secure mode within the IP network and be able to pass calls within and also to and from government voice networks.

In earlier versions of Cisco Unified CME, Cisco IOS gateways supported secure voice and data communication between legacy STE and STU devices using modem pass-through method. Cisco Unified CME 8.0 and later versions control the secure endpoints by implementing a subset of v.150.1 modem relay protocol and ensures secure communications between IP-STE endpoints and STE/STU endpoints. This allows Cisco Unified CME SCCP controlled secure endpoints to communicate with the IP-STE or legacy endpoints in secure mode.

SCCP Media Control for Secure Mode

IP-STE endpoints use the V.150.1 modem relay transport method using Future Narrow Band Digital Terminal (FNBDT) signaling over a V.32 or V.34 data pump for secure communication with other legacy STE endpoints. However, IP-STE endpoints cannot communicate with STU endpoints because STU endpoints use the modem pass-through method using a proprietary data pump and do not support the FNBDT signaling.

Secure communication between IP-STE endpoints and legacy STE endpoints support the following encryption-capable endpoints:

  • STE—Specialized encryption-capable analog or BRI phones that can communicate over V.150.1 modem relay or over modem pass-through, also known as Voice Band Data (VBD).

  • IP-STE—Specialized encryption-capable IP phones that communicate only over V.150.1 modem relay.

  • STU—Specialized encryption-capable analog phones that operate only over NSE-based modem pass-through connections.

Table 1 lists call scenarios between devices along with modem transport methods that the IP-STE endpoints use to communicate with STE endpoints.

Table 2. Supported Secure Call Scenarios and Modem Transport Methods

Device Type

STU

STE

IP-STE

STU

Pass-through

Pass-through

None

STE

Pass-through

Pass-through

Relay

IP-STE

None

Relay

Relay

Secure Communication Between STE, STU, and IP-STE Across SIP Trunk

The Secure Device Provisioning (SDP) for SIP end-to end negotiation includes four proprietary media types for secure communication between Cisco Unified CME and SIP trunk. These proprietary VBD or Modem Relay (MR) media types can be encoded into media attributes of SDP media lines. VBD capabilities are signaled using the SDP extension mechanism and Cisco proprietary nomenclature. MR capabilities are signaled through V.150.1. The following example shows VBD capabilities. The SDP syntax are based on RFC 2327 and V.150.1 Appendix E. 


a=rtpmap:100 X-NSE/8000
a=rtpmap:118 v150fw/8000
a=sqn:0
a=cdsc:1 audio RTP/AVP 118 0 18
a=cdsc: 4 audio udsprt 120
a=cpar: a=sprtmap: 120 v150mr/8000

Remote Teleworker Phones

IP phones or a Cisco IP Communicator can be connected to a Cisco Unified CME system over a WAN to support teleworkers who have offices that are remote from the Cisco Unified CME router. The maximum number of remote phones that can be supported is determined by the available bandwidth.

IP addressing is a determining factor in the most critical aspect of remote teleworker phone design. The following two scenarios represent the most common designs, the second one is the most common for small and medium businesses:

  • Remote site IP phones and the hub Cisco Unified CME router use globally routable IP addresses.

  • Remote site IP phones use NAT with unroutable private IP addresses and the hub Cisco Unified CME router uses a globally routable address (see Remote Site IP Phones Using NAT). This scenario results in one-way audio unless you use one of the following workarounds:

    • Configure static NAT mapping on the remote site router (for example, a Cisco 831 Ethernet Broadband Router) to convert between a private address and a globally routable address. This solution uses fewer Cisco Unified CME resources, but voice is unencryped across the WAN.

    • Configure an IPsec VPN tunnel between the remote site router (For example, a Cisco 831 Ethernet Broadband Router) and the Cisco Unified CME router. This solution requires Advanced IP Services or higher image on the Cisco Unified CME router if this router is used to terminate the VPN tunnel. Voice will be encrypted across the WAN. This method will also work with the Cisco VPN client on a PC to support a Cisco IP Communicator.

Figure 8. Remote Site IP Phones Using NAT

Media Termination Point for Remote Phones

Media termination point (MTP) configuration is used to ensure that Real-Time Transport Protocol (RTP) media packets from remote phones always transit through the Cisco Unified CME router. Without the MTP feature, a phone that is connected in a call with another phone in the same Cisco Unified CME system sends its media packets directly to the other phone, without the packets going through the Cisco Unified CME router. MTP forces the packets to be sourced from the Cisco Unified CME router.

When this configuration is used to instruct a phone to always send its media packets to the Cisco Unified CME router, the router acts as an MTP or proxy and forwards the packets to the destination phone. If a firewall is present, it can be 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 though they must pass through a firewall.

You must use the mtp command to explicitly enable MTP for each remote phone that sends media packets to Cisco Unified CME.

One factor to consider is whether you are using multicast music on hold (MOH) in your system. Multicast packets generally cannot be forwarded to phones that are reached over a WAN. The multicast MOH feature checks to see if MTP is enabled for a phone and if it is, MOH is not sent to that phone. If you have a WAN configuration that can forward multicast packets and you can allow RTP packets through your firewall, you can decide not to use MTP.

For configuration information, see Enable Remote Phone.

G.729r8 Codec on Remote Phones

You can select the G.729r8 codec on a remote IP phone to help save network bandwidth. The default codec is G.711 mu-law. If you use the codec g729r8 command without the dspfarm-assist keyword, the use of the G.729 codec is preserved only for calls between two phones on the Cisco Unified CME router (such as between an IP phone and another IP phone or between an IP phone and an FXS analog phone). The codec g729r8 command has no effect on a call directed through a VoIP dial peer unless the dspfarm-assist keyword is also used.

For configuration information, see Enable Remote Phone.

For information about transcoding behavior when using the G.729r8 codec, see Transcoding When a Remote Phone Uses G.729r8.

Busy Trigger and Channel Huntstop for SIP Phones

Cisco Unified CME 7.1 introduced busy trigger and huntstop channel support for SIP phones, such as the Cisco Unified IP Phone 7941G, 7941GE, 7942G, 7945G, 7961G, 7961GE, 7962G, 7965G, 7970G, 7971GE, 7975G, and 7985. For these SIP phones, the number of channels supported is limited by the amount of memory on the phone. To prevent incoming calls from overloading the phone, you can configure a busy trigger and a channel huntstop for the directory numbers on the phone.

The Channel Huntstop feature limits the number of channels available for incoming calls to a directory number. If the number of incoming calls reaches the configured limit, Cisco Unified CME does not present the next incoming call to the directory number. This reserves the remaining channels for outgoing calls or for features, such as call transfer and conferencing.

The Busy Trigger feature limits the calls to a directory number by triggering a busy response. After the number of active calls, both incoming and outgoing, reaches the configured limit, Cisco Unified CME forwards the next incoming call to the Call Forward Busy destination or rejects the call with a busy tone if Call Forward Busy is not configured.

The busy-trigger limit applies to all directory numbers on a phone. If a directory number is shared among multiple SIP phones, Cisco Unified CME presents incoming calls to those phones that have not reached their busy-trigger limit. Cisco Unified CME initiates the busy trigger for an incoming call only if all the phones sharing the directory number exceed their limit.

For configuration information, see Create Directory Numbers for SIP Phones and Assign Directory Numbers to SIP Phones.

Multiple Calls Per Line

Cisco Unified CME 9.0 provides support for the Multiple Calls Per Line (MCPL) feature on Cisco Unified 6921, 6941, 6945, and 6961 SIP IP phones and Cisco Unified 8941 and 8945 SCCP and SIP IP phones.

Before Cisco Unified CME 9.0, the maximum number of calls supported for every directory number (DN) on Cisco Unified 8941 and 8945 SCCP IP phones was restricted to two.

With Cisco Unified CME 9.0, the MCPL feature overcomes the limitation on the maximum number of calls per line.

In Cisco Unified CME 9.0, the MCPL feature is not supported on Cisco Unified 6921, 6941, 6945, and 6961 SCCP IP phones.

Cisco Unified 8941 and 8945 SCCP IP Phones

Before Cisco Unified CME 9.0, Cisco Unified 8941 and 8945 SCCP IP phones only supported two incoming calls per line and a third channel was reserved for call transfers or conference calls. These phones were also hardcoded with ephone-dn octo-line , huntstop-channel 2,max-calls -per-button 3, and busy-trigger-per-button 2.

In Cisco Unified CME 9.0, you can configure the ephone-dn dn-tag [ dual-line | octo-line] in global configuration mode and the max-calls-per-button and busy-trigger-per-button commands in ephone or ephone-template configuration mode for Cisco Unified 8941 and 8945 SCCP IP phones to configure a DN and enable the number of calls per DN, set the maximum number of calls allowed on an octo-line DN, and set the maximum number of calls allowed on an octo-line DN before activating a busy tone.

For configuration information, see Configure the Maximum Number of Calls on SCCP Phone.

Cisco Unified 6921, 6941, 6945, 6961, 8941, and 8945 SIP IP Phones

In Cisco Unified CME 9.0, the default values for the busy-trigger-per-button command is 1 for the Cisco Unified 6921, 6941, 6945, and 6961 SIP IP phones and 2 for the Cisco Unified 8941 and 8945 SIP IP phones.

You can configure the maximum number of calls before a phone receives a busy tone. For example, if you configure busy-trigger-per-button 2 in voice register pool configuration mode for a Cisco Unified 6921, 6941, 6945, or 6961 SIP IP phone, the third incoming call to the phone receives a busy tone.

For information on the Busy Trigger feature on Cisco Unified SIP IP phones, see Busy Trigger and Channel Huntstop for SIP Phones.

For configuration information, see Configure the Busy Trigger Limit on SIP Phone.

Digit Collection on SIP Phones

Digit strings dialed by phone users must be collected and matched against predefined patterns to place calls to the destination corresponding to the user's input. Before Cisco Unified CME 4.1, SIP phone users had to press the DIAL softkey or # key or wait for the interdigit-timeout to trigger call processing. In Cisco Unified CME 4.1 and later versions, two methods of collecting and matching digits are supported for SIP phones, depending on the model of phone:

Key Press Markup Language Digit Collection

Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report user input digit by digit. Each digit dialed by the phone user generates its own signaling message to Cisco Unified CME, which performs pattern recognition by matching a destination pattern to a dial peer as it collects the dialed digits. This process of relaying each digit immediately is similar to the process used by SCCP phones. It eliminates the need for the user to press the Dial softkey or wait for the interdigit timeout before the digits are sent to Cisco Unified CME for processing.

KPML is supported on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. For configuration information, see Enable KPML on a SIP Phone.

SIP Dial Plans

A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete after a user goes off-hook and dials a destination number. Dial plans allow SIP phones to perform local digit collection and recognize dial patterns as user input is collected. After a pattern is recognized, the SIP phone sends an INVITE message to Cisco Unified CME to initiate the call to the number matching the user's input. All of the digits entered by the user are presented as a block to Cisco Unified CME for processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead compared to KPML digit collection.

SIP dial plans eliminate the need for a user to press the Dial softkey or # key or to wait for the interdigit timeout to trigger an outgoing INVITE. You configure a SIP dial plan and associate the dial plan with a SIP phone. The dial plan is downloaded to the phone in the configuration file.

You can configure SIP dial plans and associate them with the following SIP phones:

  • Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE—These phones use dial plans and support KPML. If both a dial plan and KPML are enabled, the dial plan has priority.

    If a matching dial plan is not found and KPML is disabled, the user must wait for the interdigit timeout before the SIP NOTIFY message is sent to Cisco Unified CME. Unlike other SIP phones, these phones do not have a Dial softkey to indicate the end of dialing, except when on-hook dialing is used. In this case, the user can press the Dial softkey at any time to send all the dialed digits to Cisco Unified CME.

  • Cisco Unified IP Phones 7905, 7912, 7940, and 7960—These phones use dial plans and do not support KPML. If you do not configure a SIP dial plan for these phones, or if the dialed digits do not match a dial plan, the user must press the Dial softkey or wait for the interdigit timeout before digits are sent to Cisco Unified CME.

When you reset a phone, the phone requests its configuration files from the TFTP server, which builds the appropriate configuration files depending on the type of phone.

  • Cisco Unified IP Phones 7905 and 7912—The dial plan is a field in their configuration files.

  • Cisco Unified IP Phones 7911G, 7940, 7941G, 7941GE, 7960, 7961G, 7961GE, 7970G, and 7971GE—The dial plan is a separate XML file that is pointed to from the normal configuration file.

For configuration information for Cisco Unified CME, see Configure Dial Plans for SIP Phones.

Session Transport Protocol for SIP Phones

In Cisco Unified CME 4.1 and later versions, you can select TCP as the transport protocol for connecting supported SIP phones to Cisco Unified CME. Previously only UDP was supported. TCP is selected for individual SIP phones by using the session-transport command in voice register pool or voice register template configuration mode. For configuration information, see Select Session-Transport Protocol for a SIP Phone.

Real-Time Transport Protocol Call Information Display Enhancement

Before Cisco Unified CME 8.8, active RTP call information on ephone call legs were determined only by parsing the show ephone registered or show ephone offhook command output. The show voip rtp connections command showed active call information in the system but it did not apply to ephone call legs. In Cisco Unified CME 8.8 and later versions, you can display information on active RTP calls, including the ephone tag number of the phone with an active call, the channel of the ephone-dn, and the caller and called party’s numbers for the connection for both local and remote endpoints, using the show ephone rtp connections command. The output from this command provides an overview of all the connections in the system, narrowing the criteria for debugging pulse code modulation and Cisco Unified CME packets without a sniffer.


Note

When an ephone to non-ephone call is made, information on the non-ephone does not appear in a show ephone rtp connections command output. To display the non-ephone call information, use the show voip rtp connections command.

The following sample output shows all the connected ephones in the Cisco Unified CME system. The sample output shows five active ephone connections with one of the phones having the dspfarm-assist keyword configured to transcode the code on the local leg to the indicated codec. The output also shows four ephone-to-ephone calls, represented in the CallID columns of both the RTP connection source and RTP connection destination by zero values.

Normally, a phone can have only one active connection but in the presence of a whisper intercom call, a phone can have two. In the sample output, ephone-40 has two active calls: it is receiving both a normal call and a whisper intercom call. The whisper intercom call is being sent by ephone-6, which has an invalid LocalIP of 0.0.0.0. The invalid LocalIP indicates that it does not receive RTP audio because it only has a one-way voice connection to the whisper intercom call recipient. 

Router# show ephone rtp connections 
Ephone RTP active connections :
Ephone    Line  DN Chan  SrcCallID  DstCallID Codec (xcoded?)
    SrcNum DstNum  LocalIP               RemoteIP
ephone-5     1   5    1         15         14 G729 (Y)
    1005  1102  [192.168.1.100]:23192  [192.168.1.1]:2000
ephone-6     2  35    1          0          0       G711Ulaw64k (N) 
    1035  1036  [0.0.0.0]:0  [192.168.1.81]:21256
ephone-40    1 140    1          0          0       G711Ulaw64k (N)
    1140  1141  [192.168.1.81]:21244  [192.168.1.70]:20664
ephone-40    2  36    1          0          0       G711Ulaw64k (N)
    1035  1036  [192.168.1.81]:21256  [192.168.1.1]:2000
ephone-41    1 141    1          0          0       G711Ulaw64k (N)
    1140  1141  [192.168.1.70]:20664  [192.168.1.81]:21244 
Found 5 active ephone RTP connections

Ephone-Type Configuration

In Cisco Unified CME 4.3 and later versions, you can dynamically add a new phone type to your configuration without upgrading your Cisco IOS software. New phone models that do not introduce new features can easily be added to your configuration without requiring a software upgrade.

The ephone-type configuration template is a set of commands that describe the features supported by a type of phone, such as the particular phone type's device ID, number of buttons, and security support. Other phone-related settings under telephony-service, ephone-template, and ephone configuration mode can override the features set within the ephone-type template. For example, an ephone-type template can specify that a particular phone type supports security and another configuration setting can disable this feature. However, if an ephone-type template specifies that this phone does not support security, the other configuration cannot enable support for the security feature.

Cisco Unified CME uses the ephone-type template to generate XML files to provision the phone. System-defined phone types continue to be supported without using the ephone-type configuration. Cisco Unified CME checks the ephone-type against the system-defined phone types. If there is conflict with the phone type or the device ID, the configuration is rejected.

For configuration information, see Configure Ephone-Type Templates for SCCP Phones.

7926G Wireless SCCP IP Phone Support

Cisco Unified CME 8.6 adds support for the Cisco Unified 7926G Wireless SCCP IP phone. The 7926G wireless phone is phone similar to the 7925 wireless phone with a 2D barcode and EA15 module attached. The 7926G wireless phone is capable of scanning functionality. For more details on phone features and functionality, see Cisco Unified IP Phone 7900 Series User Guide.

Cisco Unified CME 8.6 supports the scanning function on the 7926G SCCP wireless phone using the ephone built-in device type. Table 1 shows supported values for the ephone-type for 7926G wireless phone.

Table 3. Supported Values for Ephone-Type Command

Supported Device

device-id

device-type

num-buttons

max-presentation

Cisco Unified Wireless IP Phone 7926G

577

7926

6

2

To support service provisioning, an XML file is constructed externally and applied to the ephone-template of the phone. To allow the phone to read the external XML file, you are required to create-cnf and download the XML file to the ephone. For more information on configuring PhoneServices XML file, see Configure Phone Services XML File for Cisco Unified Wireless Phone 7926G.

The following is an example of the <phoneServices> XML file: 


		<phoneServices  useHTTPS="true">  
		<provisioning>0</provisioning>  
		<phoneService  type="1" category="0">  
		<name>Missed Calls</name> 
		<url>Application:Cisco/MissedCalls</url>  
		<vendor></vendor>  
		<version></version>  
		</phoneService>  
		<phoneService  type="0" category="1">  
		<displayName>Store Ops</displayName>  
		<name>Store Ops</name>  
		<url>http://1.4.206.105/Midlets/StoreOps.jad?StoreNumber=1777</url>  
		<http://1.4.206.105/Midlets/StoreOps.jad?StoreNumber=1777%3c/url%3e>  
		<http://1.4.206.105/Midlets/StoreOps.jad?StoreNumber=1777%3c/url%3e>  
		<vendor>CiscoSystems</vendor>  
		<version>0.0.82</version>  
		</phoneService>  
		</phoneServices>

Enhanced Line Mode

Enhanced Line Mode allows you to use the buttons on both sides of the phone screen to configure Line Keys (DNs), Feature Buttons, or Speed Dial.

In a scenario where you have Line Keys, Feature Button, and Speed Dial configured under voice register pool configuration mode for phones that are supported on Unified CME, the priority is set as follows:

  • Line Keys

  • Speed Dial

  • Feature Button

From Unified CME Release 12.3, support is introduced for Enhanced Line Mode (ELM) on Cisco IP Phone 8800 Series. The support is introduced for all Cisco IP Phone 8800 Series phones, except Cisco Wireless IP Phone 8821, Cisco Unified IP Conference Phone 8831, and Cisco IP Conference Phone 8832. ELM for Unified CME is supported on the Cisco 4000 Series Integrated Services Routers. For Cisco IP Phone 8800 Series, a maximum of 10 phone buttons can be configured for ELM lines.

For ELM on Unified CME, you need to configure the CLI command service phone lineMode 1 under telephony-service configuration mode to enable Enhanced Line Mode on phones. The Cisco IP Phone 8800 Series configured on Unified CME uses the vendor config XML body in the CNF file to verify if the CLI command service phone lineMode 1 is added to enable ELM mode. For a sample configuration of ELM on Unified CME, see Example for Configuring Enhanced Line Mode on Unified CME.


Note

The CLI command service phone lineMode is case-sensitive, and must be entered exactly as mentioned.

You can enable ELM on Unified CME using the CLI command service phone lineMode as follows: 


Router(config)#telephony-service
Router(config-telephony)#service phone lineMode ?
  WORD  enter the phone xml file parameter text for the previously entered
        parameter name
Router(config-telephony)#service phone lineMode 1
Router(config-telephony)#create cnf-files
Router(config-telephony)#end

Once you enable service phone lineMode 1 under telephony-service for ELM, you need to create profile and restart the phones under voice register global configuration mode to enable ELM for the Cisco IP Phone 8800 series phones on Unified CME. 


Router(config)#voice register global
Router(config-register-global)#create profile
Router(config-register-global)#restart
Router(config-register-global)#end

Feature Support on Enhanced Line Mode

The following features are supported for ELM on Cisco IP Phone 8800 Series:

  • HLog

  • DND

  • Park

  • Redial

  • Mobility

  • Group Pickup

  • Meet Me

  • Mobility

  • Pickup

  • Privacy

KEM Support for Cisco Unified SIP IP Phones

For Unified CME 12.3 and prior releases, KEM support is limited to C-KEM and BE-KEM device types. From Unified CME Release 12.5, Key Expansion Module (KEM) device types A-KEM (Audio) and V-KEM (Video) are supported for Cisco IP Phone 8800 Series. The support is introduced for both SLM (Session Line Mode) and ELM (Enhanced Line Mode) configuration. You can switch from SLM to ELM mode to use buttons on both sides of the Cisco IP Phone 8800 Series.

The following endpoints are supported as part of Unified CME Release 12.5:

  • Cisco IP Phone 8851—Supports upto 2 A-KEM Modules.

  • Cisco IP Phone 8851NR—Supports upto 2 A-KEM Modules

  • Cisco IP Phone 8861—Supports upto 3 A-KEM Modules.

  • Cisco IP Phone 8865—Supports upto 3 V-KEM Modules

An A-KEM or V-KEM Module supports a maximum of 28 lines. Hence, the total number of lines on the supported phone types for Unified CME 12.5 are as follows:

Table 4. A-KEM and V-KEM Line Support

Phone Model

Number of KEM Lines Supported

Line Support (With SLM)

Line Support (With ELM)

8851

56 (2*28)

61 (56+5)

66 (56+10)

8851NR

56(2*28)

61 (56+5)

66 (56+10)

8861

84 (3*28)

89 (84+5)

94 (84+10)

8865

84 (3*28)

89 (84+5)

94 (84+10)

V-KEM is supported only with the 8865 phone type. You need to configure CP-8800-Video to support V-KEM with 8865 phones. You need to configure CP-8800-Audio to support A-KEM with the phone types 8851, 8851NR, and 8861. The phone types 8851, 8851NR, and 8861 also support CKEM and BEKEM.


Note

A mixed deployment of KEM Modules is not supported for any phone type. For example, if the phone type 8861 supports three KEM modules, then all three KEM modules have to be either CKEM, BEKEM, or CP-8800-Audio.

To enable A-KEM or V-KEM on Unified CME, you need to configure the KEM option for the phone type under voice register pool configuration mode for Unified CME 12.5 and later releases: 

Router(config)# enable
Router(config)# configure terminal
Router(config)# voice register pool
Router(config-register-pool)# type 8851 addon 1 CP-8800-Audio 2 CP-8800-Audio
Router(config-register-pool)# type 8851NR addon 1 CP-8800-Audio 2 CP-8800-Audio
Router(config-register-pool)# type 8861 addon 1 CP-8800-Audio 2 CP-8800-Audio 3 CP-8800-Audio
Router(config-register-pool)# type 8865 addon 1 CP-8800-Video 2 CP-8800-Video 3 CP-8800-Video

To configure KEM on Unified SIP Phones, see Configure KEMs on SIP Phones.

For more information on the KEM support for Cisco Unified 8851/51NR, 8861, 8865, 8961, 9951, and 9971 SIP IP Phones, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified Secure SRST.

Key Mapping

The mapping of configured keys on a phone depends on the number of KEMs attached to the phone.

If only one CKEM is attached to a phone and the number of keys configured is 114, only 36 keys on the CKEM are mapped to the configured keys on the phone. The rest of the keys are not visible on the phone or the KEM. The maximum number of supported keys on each A-KEM and V-KEM device is 28. For information on A-KEM and V-KEM support, see Table 1.

Call Control

All call control features are supported by KEMs on Cisco Unified SIP IP phones. Any feature that can be configured on the phone keys can also be configured on the KEM.

XML Updates

  • There is no separate firmware for KEMs, instead they are built in as part of the phones.

  • The number of XML entries in the configuration file increases with the number of keys configured.

  • The device type for KEMs is C-KEM, BE-KEM, A-KEM, and V-KEM. The maximum number of supported keys on each C-KEM device is 36. The maximum number of supported keys on each A-KEM and V-KEM device is 28.

Restrictions for KEM Support

  • KEMs are not supported for Cisco Unified SCCP IP phones and Cisco Unified SIP IP phones other than the Cisco Unified 8851/51NR, 8861, 8865, 8961, 9951, and 9971 SIP IP phones.

  • Features configured on keys are disabled when supported Cisco Unified SIP IP phones are in Cisco Unified SIP SRST.

  • All Cisco Unified 8851/51NR, 8861, 8865, 8961, 9951, and 9971 SIP IP phone restrictions and limitations apply to KEMs.

  • All Cisco Unified CME and Cisco Unified SIP SRST feature restrictions and limitations apply to KEMs.

For more information on how the blf-speed-dial , number , and speed-dial commands, in voice register pool configuration mode, have been modified, see Cisco Unified Communications Manager Express Command Reference.

For information on installing KEMs on Cisco Unified IP Phone, see “Installing a Key Expansion Module on the Cisco Unified IP Phone” section of Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 10.0 .

For information on installing KEMs on Cisco Unified 8811, 8841, 8851, 8851NR, 8865, and 8861 Phones, see Cisco IP Phone Key Expansion Module section of Cisco IP Phone 8800 Series Administration Guide for Cisco Unified Communications Manager.

Fast-Track Configuration Approach for Cisco Unified SIP IP Phones

In Cisco Unified CME Release 10.0, the Fast-Track Configuration feature provides a new configuration utility using which you can input the phone characteristics of a new SIP phone model. This utility allows you to configure the existing SIP line features to the new SIP phone models. In the fast-track configuration, an option is provided to input an existing SIP phone as a reference phone. This feature is supported only on new SIP phone models that do not need any changes in the software protocols and the Cisco Unified CME application.


Note

To deploy Cisco Unified SIP IP phones on Cisco Unified CME using the fast-track configuration approach, you require Cisco IOS Release 15.3(3)M or a later release.

Forward Compatibility

When a new SIP phone model is configured using the fast-track configuration approach. and the Cisco Unified CME is upgraded to a later version that supports the new SIP phone model, the fast-track configuration pertaining to that SIP phone model is removed automatically. If the Cisco Unified CME is downgraded to a version that does not have the built-in support, the fast-track configuration should be applied again.

To support Fast-Track Configuration feature, the voice register pool-type command has been introduced in the global configuration mode. The properties of the new SIP phone can be configured under the voice register pool-type submode. In addition to the explicit configuration of the phone’s properties, the reference-pooltype option can be used to inherit the properties of an existing SIP phone.

Localization support

CME supports localization for phones in fast-track mode through locale installer. However, the locale package should have .jar files for a specific phone model to make the feature work.

To use the locale installer, see Locale Installer for Cisco Unified SIP IP Phones .

For new SIP phone models validated using Fast-track configuration and the supported locale package version, see Phone Feature Support Guide for Unified CME, Unified SRST, Unified E-SRST, and Unified Secure SRST.

Restrictions for Fast-Track Support

  • The fast-track configuration does not allow you to use the following phone models as reference phone:

    • ATA—Cisco ATA-186 and Cisco ATA-188

    • 7905—Cisco Unified IP Phone 7905 and Cisco Unified IP Phone 7905G

    • 7912—Cisco Unified IP Phone 7912 and Cisco Unified IP Phone 7912G

    • 7940—Cisco Unified IP Phone 7940 and Cisco Unified IP Phone 7940G

    • 7960—Cisco Unified IP Phone 7960 and Cisco Unified IP Phone 7960G

    • P100—PingTel Xpressa 100

    • P600—Polycom SoundPoint IP 600

    • Existing Cisco Unified SIP IP phones are not allowed to be configured as new Cisco Unified SIP IP phones using the fast-track configuration approach.

    • The reference-pooltype functionality is allowed only on existing SIP phone models. New SIP phone models configured using the fast-track configuration approach cannot be used as a reference phone.

    • The fast-track configuration approach supports only the XML format and not support the text format for phone configuration.

    • The fast-track approach does not support the new SIP phone models that have a new call flow, new message flow, or a new configuration file format that are not supported by the Cisco Unified CME.

For configuration information, see Provision SIP Phones to Use the Fast-Track Configuration Approach.

For configuration examples, see Example for Fast-Track Configuration Approach.

Configure Phones for a PBX System

This section contains the following tasks:

Create Directory Numbers for SCCP Phones

To create a directory number in Cisco Unified CME for a SCCP phone, intercom line, voice port, or a message-waiting indicator (MWI), perform the following steps for each directory number to be created. Each ephone-dn becomes a virtual line, or extension, on which call connections can be made. Each ephone-dn configuration automatically creates one or more virtual dial peers and virtual voice ports to make those call connections.


Note
To create and assign directory numbers to be included in an overlay set, see Configure Overlaid Ephone-dns on SCCP Phones.

Restriction

  • The Cisco Unified IP Phone 7931G is a SCCP keyset phone and, when configured for a key system, does not support the dual-line option for a directory number. To configure a Cisco Unified IP Phone 7931G, see Configure Phones for a Key System.

  • Octo-line directory numbers are not supported by the Cisco Unified IP Phone 7902, 7920, or 7931, or by analog phones connected to the Cisco VG224 or Cisco ATA.

  • Octo-line directory numbers are not supported in button overlay sets.

  • Octo-line directory numbers do not support the trunk command.

Before you begin

  • Maximum number of directory numbers must be changed from the default of 0 by using the max-dn command.

  • Octo-line directory numbers are supported in Cisco Unified CME 4.3 and later versions.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone-dn dn-tag [ dual-line | octo-line]
  4. number number [ secondary number] [ no-reg [ both | primary] ]
  5. huntstop [ channel number]
  6. name name
  7. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

ephone-dn dn-tag [ dual-line | octo-line]

Example:

Router(config)# ephone-dn 7 octo-line

Enters ephone-dn configuration mode to create a directory number for a SCCP phone.

  • dual-line —(Optional) Enables two calls per directory number. Supports features such as call waiting, call transfer, and conferencing with a single ephone-dn.

  • octo-line —(Optional) Enables eight calls per directory number. Supported in Cisco Unified CME 4.3 and later versions.

  • To change the line mode of a directory number, for example from dual-line to octo-line or the reverse, you must first delete the ephone-dn and then recreate it.

Step 4

number number [ secondary number] [ no-reg [ both | primary] ]

Example:

Router(config-ephone-dn)# number 2001

Configures an extension number for this directory number.

  • Configuring a secondary number supports features such as call waiting, call transfer, and conferencing with a single ephone-dn.

Step 5

huntstop [ channel number]

Example:

Router(config-ephone-dn)# huntstop channel 4

(Optional) Enables Channel Huntstop, which keeps a call from hunting to the next channel of a directory number if the first channel is busy or does not answer.

  • channel number —Number of channels available to accept incoming calls. Remaining channels are reserved for outgoing calls and features such as call transfer, call waiting, and conferencing. Range: 1 to 8. Default: 8.

  • number argument is supported for octo-line directory numbers only.

Step 6

name name

Example:

Router(config-ephone-dn)# name Smith, John

(Optional) Associates a name with this directory number.

  • Name is used for caller-ID displays and in the local directory listings.

  • Must follow the name order that is specified with the directory command.

Step 7

end

Example:

Router(config-ephone-dn)# end

Returns to privileged EXEC mode.

Example

Example for Nonshared Octo-Line Directory Number

In the following example, ephone-dn 7 is assigned to phone 10 and not shared by any other phone. There are two active calls on ephone-dn 7. Because the busy-trigger-per-button command is set to 2, a third incoming call to extension 2001 is either rejected with a busy tone or forwarded to another destination if Call Forward Busy is configured. The phone user can still make an outgoing call or transfer or conference a call on ephone-dn 7 because the max-calls-per-button command is set to 3, which allows a total of three calls on ephone-dn 7. 


ephone-dn  7  octo-line
 number 2001 
	name Smith, John 
	huntstop channel 4
! 
! 
ephone  10
 max-calls-per-button 3
 busy-trigger-per-button 2 
 mac-address 00E1.CB13.0395 
 type 7960 
 button  1:7

Example for Shared Octo-Line Directory Number

In the following example, ephone-dn 7 is shared between phone 10 and phone 11. There are two active calls on ephone-dn 7. A third incoming call to ephone-dn 7 rings only phone 11 because its busy-trigger-per-button command is set to 3. Phone 10 allows a total of three calls, but it rejects the third incoming call because its busy-trigger-per-button command is set to 2. A fourth incoming call to ephone-dn 7 on ephone 11 is either rejected with a busy tone or forwarded to another destination if Call Forward Busy is configured. The phone user can still make an outgoing call or transfer or conference a call on ephone-dn 7 on phone 11 because the max-calls-per-button command is set to 4, which allows a total of four calls on ephone-dn 7 on phone 11. 


ephone-dn  7  octo-line 
 number 2001 
 name Smith, John 
 huntstop channel 4 
! 
! 
ephone  10 
 max-calls-per-button 3 
 busy-trigger-per-button 2 
 mac-address 00E1.CB13.0395> 
 type 7960
 button  1:7 
! 
! 
! 
ephone  11 
 max-calls-per-button 4 
 busy-trigger-per-button 3 
 mac-address 0016.9DEF.1A70
 type 7960 
 button  1:7

What to do next

After creating directory numbers, you can assign one or more directory numbers to a Cisco Unified IP Phone. See Assign Directory Numbers to SCCP Phones.

Configure Ephone-Type Templates for SCCP Phones


Restriction
Ephone-type templates are not supported for system-defined phone types. For a list of system-defined phone types, see the type command in Cisco Unified CME Command Reference.

Before you begin

Cisco Unified CME 4.3 or a later version.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone-type phone-type [ addon]
  4. device-id number
  5. device-name name
  6. device-type phone-type
  7. num-buttons number
  8. max-presentation number
  9. addon
  10. security
  11. phoneload
  12. utf8
  13. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

ephone-type phone-type [ addon]

Example:

Router(config)# ephone-type E61

Enters ephone-type configuration mode to create an ephone-type template.

  • phone-type —Unique label that identifies the type of IP phone for which the phone-type template is being defined.

  • addon —(Optional) Phone type is an add-on module, such as the Cisco Unified IP Phone 7915 Expansion Module.

Step 4

device-id number

Example:

Router(config-ephone-type)# device-id 376

Specifies the device ID for the phone type.

  • This device ID must match the predefined device ID for the specific phone model.

  • If this command is set to the default value of 0, the ephone-type is invalid.

  • See Table 1 for a list of supported device IDs.

Step 5

device-name name

Example:

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

Assigns a name to the phone type.

  • See Table 1 for a list of supported device types.

Step 6

device-type phone-type

Example:

Router(config-ephone-type)# device-type E61

Specifies the device type for the phone.

Step 7

num-buttons number

Example:

Router(config-ephone-type)# num-buttons 1

Number of line buttons supported by the phone type.

  • number —Range: 1 to 100. Default: 0.

  • See Table 1 for the number of buttons supported by each phone type.

Step 8

max-presentation number

Example:

Router(config-ephone-type)# max-presentation 1

Number of call presentation lines supported by the phone type.

  • number —Range: 1 to 100. Default: 0.

  • See Table 1 for the number of presentation lines supported by each phone type.

Step 9

addon

Example:

Router(config-ephone-type)# addon

(Optional) Specifies that this phone type supports an add-on module, such as the Cisco Unified IP Phone 7915 Expansion Module.

Step 10

security

Example:

Router(config-ephone-type)# security

(Optional) Specifies that this phone type supports security features.

  • This command is enabled by default.

Step 11

phoneload

Example:

Router(config-ephone-type)# phoneload

(Optional) Specifies that this phone type requires that the load command be configured.

  • This command is enabled by default.

Step 12

utf8

Example:

Router(config-ephone-type)# utf8

(Optional) Specifies that this phone type supports UTF8.

  • This command is enabled by default.

Step 13

end

Example:

Router(config-ephone-type)# end

Exits to privileged EXEC mode.

Ephone-Type Parameters for Supported Phone Types

Table 1 lists the required device ID, device type, and the maximum number of buttons and call presentation lines that are supported for each phone type that can be added with ephone-type templates.

Table 5. Supported Values for Ephone-Type Commands

Supported Device

device-id

device-type

num-buttons

max-presentation

Cisco Unified IP  Phoone 6901

547

6901

1

1

Cisco Unified IP  Phone 6911

548

6911

10

1

Cisco Unified IP Phone 6945

564

6945

4

2

Cisco Unified IP Phone 7915 Expansion Module with 12 buttons

227

7915

12

0 (default)

Cisco Unified IP  Phone 7915 Expansion Module with 24 buttons

228

7915

24

0

Cisco Unified IP  Phone 7916 Expansion Module with 12 buttons

229

7916

12

0

Cisco Unified IP  Phone 7916 Expansion Module with 24 buttons

230

7916

24

0

Cisco Unified  Wireless IP Phone  7925

484

7925

6

4

Cisco Unified IP Conference  Station 7937G

431

7937

1

6

Cisco Unified IP Phone 8941

586

8941

4

3

Cisco Unified IP Phone 8945

585

8945

4

3

Cisco Unified IP Phone 8941 with Fast-Track configuration support

586

8941

4

3

Cisco Unified IP Phone 8945 with Fast-Track configuration support

586

8945

4

3

Nokia E61

376

E61

1

1

Example

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
		!
		ephone  2
		 mac-address 001C.821C.ED23
		 type E61
		 button  1:2

Assign Directory Numbers to SCCP Phones

This task sets up the initial ephone-dn-to-ephone relationships: how and which extensions appear on each phone. To create and modify phone-specific parameters for individual SCCP phones, perform the following steps for each SCCP phone to be connected in Cisco Unified CME.


Note

To create and assign directory numbers to be included in an overlay set, see Configure Overlaid Ephone-dns on SCCP Phones.


Restriction

  • For Watch mode. If the watched directory number is associated with several phones, then the watched phone is the one on which the watched directory number is on button 1 or the one on which the watched directory number is on the button that is configured by using theauto-line command, with auto-line having priority. For configuration information, see Automatic Line Selection.

  • Octo-line directory numbers are not supported by the Cisco Unified IP Phone 7902, 7920, or 7931, or by analog phones connected to the Cisco VG224 or Cisco ATA.

  • Octo-line directory numbers are not supported in button overlay sets.

Before you begin

  • To configure a phone line for Watch (w) mode by using the button command, Cisco Unified CME 4.1 or a later version.

  • To configure a phone line for Monitor (m) mode by using the button command, Cisco CME 3.0 or a later version.

  • To assign a user-defined phone type in Cisco Unified CME 4.3 or a later version, you must first create an ephone-type template. See Configure Ephone-Type Templates for SCCP Phones.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone phone-tag
  4. mac-address [ mac-address]
  5. type phone-type [ addon 1 module-type [ 2 module-type] ]
  6. button button-number { separator} dn-tag [ , dn-tag...] [ button-number { x} overlay-button-number] [ button-number...]
  7. max-calls-per-button number
  8. busy-trigger-per-button number
  9. keypad-normalize
  10. nte-end-digit-delay [ milliseconds]
  11. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

ephone phone-tag

Example:

Router(config)#ephone 6

Enters ephone configuration mode.

  • phone-tag —Unique sequence number that identifies this ephone during configuration tasks. The maximum number of ephones is version and platform-specific. Type ? to display range.

Step 4

mac-address [ mac-address]

Example:

Router(config-ephone)#mac-address 2946.3f2.311

Specifies the MAC address of the IP phone that is being configured.

  • mac-address —(Optional) For CiscoUnifiedCME 3.0 and later versions, it is not required to register phones before configuring the phone because CiscoUnifiedCME can detect MAC addresses and automatically populate phone configurations with the MAC addresses and phone types for individual phones. Not supported for voice-mail ports.

Step 5

type phone-type [ addon 1 module-type [ 2 module-type] ]

Example:

Router(config-ephone)# type 7960 addon 1 7914

Specifies the type of phone.

  • CiscoUnifiedCME 4.0 and later versionsThe only types to which you can apply an add-on module are 7960 , 7961 ,7961GE , and 7970 .

  • CiscoCME 3.4 and earlier versionsThe only type to which you can apply an add-on module is 7960 .

Step 6

button button-number { separator} dn-tag [ , dn-tag...] [ button-number { x} overlay-button-number] [ button-number...]

Example:

Router(config-ephone)# button 1:10 2:11 3b12 4o13,14,15

Associates a button number and line characteristics with an extension (ephone-dn). Maximum number of buttons is determined by phone type.

Note

 

The CiscoUnified IPPhone7910 has only one line button but can be given two ephone-dn tags.

Step 7

max-calls-per-button number

Example:

Router(config-ephone)# max-calls-per-button 3

(Optional) Sets the maximum number of calls, incoming and outgoing, allowed on an octo-line directory number on this phone.

  • number —Range: 1 to 8. Default: 8.

  • This command is supported in CiscoUnifiedCME4.3 and later versions.

  • 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.

  • This command can also be configured in ephone-template configuration mode and applied to one or more phones. The ephone configuration has priority over the ephone-template configuration.

Step 8

busy-trigger-per-button number

Example:

Router(config-ephone)# busy-trigger-per-button 2

(Optional) Sets the maximum number of calls allowed on this phones octo-line directory numbers before triggering Call Forward Busy or a busy tone.

  • number —Range: 1 to 8. Default: 0 (disabled).

  • This command is supported in CiscoUnifiedCME4.3 and later versions.

  • After the number of existing calls, incoming and outgoing, on an octo-line directory number exceeds the number of calls set with this command, the next incoming call to the directory number is forwarded to the Call Forward Busy destination if configured, or the call is rejected with a busy tone.

  • This command must be set to a value that is less than or equal to the value set with the max-calls-per-button command.

  • This command can also be configured in ephone-template configuration mode and applied to one or more phones. The ephone configuration has priority over the ephone-template configuration.

Step 9

keypad-normalize

Example:

Router(config-ephone)# keypad-normalize

(Optional) Imposes a 200-millisecond delay before each keypad message from an IP phone.

  • When used with the nte-end-digit-delay command, this command ensures that the delay configured for a dtmf-end event is always honored.

Step 10

nte-end-digit-delay [ milliseconds]

Example:

Router(config-ephone)# nte-end-digit-delay 150

(Optional) Specifies the amount of time that each digit in the RTP NTE end event in an RFC2833 packet is delayed before being sent.

  • This command is supported in CiscoUnifiedCME 4.3 and later versions.

  • milliseconds —length of delay. Range: 10 to 200. Default: 200.

  • To enable the delay, you must also configure the dtmf-interworking rtp-nte command in voice-service or dial-peer configuration mode. For information, see Enable DTMF Integration Using RFC 2833.

  • This command can also be configured in ephone-template configuration mode. The value set in ephone configuration mode has priority over the value set in ephone-template mode.

Step 11

end

Example:

Router(config-ephone)# end

Returns to privileged EXEC mode.

Example

Example for assigning directory number to SCCP Phone

The following example assigns extension 2225 in the Accounting Department to button 1 on ephone 2: 


			ephone-dn 25
			 number 2225
			 name Accounting 
			
			 ephone 2 
			 mac-address 00E1.CB13.0395 
			 type 7960
			 button 1:25

What to do next

Create Directory Numbers for SIP Phones

To create a directory number in Cisco Unified CME for a SIP phone, intercom line, voice port, or a message-waiting indicator (MWI), perform the following steps for each directory number to be created.


Restriction

  • Valid characters in voice register DN include 0–9, '.', '+', '*', and '#'.

  • The name or label that is associated with a directory number that is configured under voice register dn or voice register global configuration mode cannot contain special characters such as quotes ("), angle brackets (<, >), ampersand (&), and percentage (%).

  • To allow insertion of '#' at any place in voice register DN, the CLI "allow-hash-in-dn" is configured in voice register global mode.

  • When the CLI "allow-hash-in-dn" is configured, the user is required to change the dial-peer terminator from '#' (default terminator) to another valid terminator in configuration mode. The other terminators that are supported include '0'-'9', 'A'-'F', and '*'.

  • Maximum number of directory numbers that are supported by a router is version and platform dependent.

  • Call Forward All, Presence, and message-waiting indication (MWI) features in Cisco Unified CME 4.1 and later versions require that SIP phones be configured with a directory number using the dn keyword with the number command; direct line numbers are not supported.

  • SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only.

  • The Media Flow-around feature configured with the media flow-around command is not supported by Cisco Unified CME with SIP phones.

  • SIP shared-line directory numbers are not supported by the Cisco Unified IP Phone 7902, 7920, 7931, 7940, or 7960, or by analog phones connected to the Cisco VG224.

  • For Unified CME 12.1 and prior releases, SIP shared-line directory numbers cannot be members of voice hunt groups.

  • If this directory number is used as shared line, you can associate the directory number to a maximum of 16 phones.

Before you begin

  • Cisco CME 3.4 or a later version.

  • SIP shared-line directory numbers are supported in Cisco Unified CME 7.1 and later versions.

  • registrar server command must be configured. For configuration information, see Enable Calls in Your VoIP Network.

  • In Cisco Unified CME 7.1 and later versions, the maximum number of directory numbers must be changed from the default of 0 by using the max-dn (voice register global) command. For configuration information, see Set Up Cisco Unified CME for SIP Phones.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice register dn dn-tag
  4. number number
  5. shared-line [ max-calls number-of-calls]
  6. huntstop channel number-of-channels
  7. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

voice register dn dn-tag

Example:

Router(config)# voice register dn 17

Enters voice register DN configuration mode to define a directory number for a SIP phone, intercom line, voice port, or a message-waiting indicator (MWI).

Step 4

number number

Example:

Router(config-register-dn)# number 7001

Defines a valid number for a directory number.

Step 5

shared-line [ max-calls number-of-calls]

Example:

Router(config-register-dn)# shared-line max-calls 6

(Optional) Creates a shared-line directory number.

  • max-calls number-of-calls (Optional)—Maximum number of calls, both incoming and outgoing. Range: 2–16. Default: 2.

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

  • This command is supported in Cisco Unified CME 7.1 and later versions.

Step 6

huntstop channel number-of-channels

Example:

Router(config-register-dn)# huntstop channel 3

(Optional) Enables Channel Huntstop, which keeps a call from hunting to the next channel of a directory number if the first channel is busy or does not answer.

  • number-of-channels —Number of channels available to accept incoming calls on the directory number. Remaining channels are reserved for outgoing calls and features, such as Call Transfer, Call Waiting, and Conferencing. Range: 1–50. Default: 0 (disabled).

  • This command is supported in Cisco Unified CME 7.1 and later versions.

Step 7

end

Example:

Router(config-register-dn)# end

Exits to privileged EXEC mode.

Example

Example for assigning directory numbers to SIP Phones

The following example shows directory number 24 configured as a shared line and assigned to phone 124 and phone 125: 


				voice register dn  24 
			  number 8124
			  shared-line max-calls 6
			 !
			 voice register pool  124 
			  id mac 0017.E033.0284 
			  type 7965
			  number 1 dn 24 
			 ! 
			 voice register pool  125
			  id mac 00E1.CB13.0395 
			  type 7965
			 number 1 dn 24

Assign Directory Numbers to SIP Phones

This task sets up which extensions appear on each phone. To create and modify phone-specific parameters for individual SIP phones, perform the following steps for each SIP phone to be connected in Cisco Unified CME.


Note

If your Cisco Unified CME system supports SCCP and SIP phones, do not connect your SIP phones to your network until after you have verified the configuration profile for the SIP phone.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice register pool pool-tag
  4. id { network address mask mask | ip address mask mask | mac address}
  5. type phone-type
  6. number tag dn dn-tag
  7. busy-trigger-per-button number-of-calls
  8. username username password password
  9. dtmf-relay { [ cisco-rtp] [ rtp-nte] [ sip-notify] }
  10. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

voice register pool pool-tag

Example:

Router(config)# voice register pool 3

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.

Step 4

id { network address mask mask | ip address mask mask | mac address}

Example:

Router(config-register-pool)#
			 id mac 0009.A3D4.1234

Explicitly identifies a locally available individual SIP phone to support a degree of authentication.

Step 5

type phone-type

Example:

Router(config-register-pool)# 
			 type  7960-7940

Defines a phone type for the SIP phone being configured.

Step 6

number tag dn dn-tag

Example:

Router(config-register-pool)# number 1 dn 17

Associates a directory number with the SIP phone being configured.

  • dn dn-tag —identifies the directory number for this SIP phone as defined by the voice register dn command.

Step 7

busy-trigger-per-button number-of-calls

Example:

Router(config-register-pool)# busy-trigger-per-button 2

(Optional) Sets the maximum number of calls allowed on any of this phone�s directory numbers before triggering Call Forward Busy or a busy tone.

  • number-of-calls —Maximum number of calls allowed before Cisco Unified CME forwards the next incoming call to the Call Forward Busy destination, if configured, or rejects the call with a busy tone. Range: 1 to 50.

  • This command is supported in Cisco Unified CME 7.1 and later versions.

Step 8

username username password password

Example:

Router(config-register-pool)# username smith password 123zyx

(Optional) Required only if authentication is enabled with the authenticate command. Creates an authentication credential.

Note

 

This command is not for SIP proxy registration. The password will not be encrypted. All lines in a phone will share the same credential.

  • username —identifies a local Cisco Unified IP phone user. Default: Admin.

Step 9

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

Example:

Router(config-register-pool)# dtmf-relay rtp-nte

(Optional) Specifies a list of DTMF relay methods that can be used by the SIP phone to relay DTMF tones.

Note

 

SIP phones natively support in-band DTMF relay as specified in RFC 2833.

Step 10

end

Example:

Router(config-register-pool)# end

Returns to privileged EXEC mode.

Example for configuring SIP Nonshared Line

In the following example, voice register dn 23 is assigned to phone 123. The fourth incoming call to extension 8123 is not presented to the phone because the huntstop channel command is set to 3. Because the busy-trigger-per-button command is set to 2 on phone 123 and Call Forward Busy is configured, the third incoming call to extension 8123 is forwarded to extension 8200. 


voice register dn  23
 number 8123
 call-forward b2bua busy 8200
 huntstop channel 3
!
voice register pool  123
 busy-trigger-per-button 2
 id mac 0009.A3D4.1234
 type 7965
 number 1 dn 23

In the following example, voice register dn 24 is shared by phones 124 and 125. The first two incoming calls to extension 8124 ring both phones. A third incoming call rings only phone 125 because its busy-trigger-per-button command is set to 3. The fourth incoming call to extension 8124 triggers Call Forward Busy because the busy trigger limit on all phones is exceeded. 


voice register dn  24
 number 8124
 call-forward b2bua busy 8200
 shared-line max-calls 6
 huntstop channel 6
!
voice register pool  124
 busy-trigger-per-button 2
 id mac 0017.E033.0284
 type 7965
 number 1 dn 24
!
voice register pool  125
 busy-trigger-per-button 3
 id mac 00E1.CB13.0395
 type 7965
 number 1 dn 24

What to do next

Configure Dial Plans for SIP Phones

Dial plans enable SIP phones to recognize digit strings dialed by users. After the phone recognizes a dial pattern, it automatically sends a SIP INVITE message to the Cisco Unified CME to initiate the call and does not require the user to press the Dial key or wait for the interdigit timeout. To define a dial plan for a SIP phone, perform the following steps.

Before you begin

  • Cisco Unified CME 4.1 or a later version.

  • mode cme command must be enabled in Cisco Unified CME.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice register dialplan dialplan-tag
  4. type phone-type
  5. pattern tag string [ button button-number] [ timeout seconds] [ user { ip | phone} ] or filename filename
  6. exit
  7. voice register pool pool-tag
  8. dialplan dialplan-tag
  9. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

voice register dialplan dialplan-tag

Example:

Router(config)# voice register dialplan 1

Enters voice register dialplan configuration mode to define a dial plan for SIP phones.

Step 4

type phone-type

Example:

Router(config-register-dialplan)# type 7905-7912

Defines a phone type for the SIP dial plan.

  • 7905-7912 —Cisco Unified IP Phone 7905, 7905G, 7912, or 7912G.

  • 7940-7960-others —Cisco Unified IP Phone 7911, 7940, 7940G, 7941, 7941GE, 7960, 7960G, 7961, 7961GE, 7970, or 7971.

  • The phone type specified with this command must match the type of phone for which the dial plan is used. If this phone type does not match the type assigned to the phone with the type command in voice register pool mode, the dial-plan configuration file is not generated.

  • You must enter this command before using the pattern or filename command in the next step.

Step 5

pattern tag string [ button button-number] [ timeout seconds] [ user { ip | phone} ] or filename filename

Example:

Router(config-register-dialplan)# pattern 1 52...

or 

Router(config-register-dialplan)# filename dialsip

Defines a dial pattern for a SIP dial plan.

  • tag —Number that identifies the dial pattern. Range: 1 to 24.

  • string —Dial pattern, such as the area code, prefix, and first one or two digits of the telephone number, plus wildcard characters or dots (.) for the remainder of the dialed digits.

  • button button-number —(Optional) Button to which the dial pattern applies.

  • timeout seconds — (Optional) Time, in seconds, that the system waits before dialing the number entered by the user. Range: 0 to 30. To have the number dialed immediately, specify 0. If you do not use this parameter, the phone's default interdigit timeout value is used (10 seconds).

  • user —(Optional) Tag that automatically gets added to the dialed number. Do not use this keyword if Cisco Unified CME is the only SIP call agent.

  • ip —Uses the IP address of the user.

  • phone —Uses the phone number of the user.

  • Repeat this command for each pattern that you want to include in this dial plan.

or

Specifies a custom XML file that contains the dial patterns to use for the SIP dial plan.

  • You must load the custom XML file must into flash and the filename cannot include the .xml extension.

  • The filename command is not supported for the Cisco Unified IP Phone 7905 or 7912.

Step 6

exit

Example:

Router(config-register-dialplan)# exit

Exits dialplan configuration mode.

Step 7

voice register pool pool-tag

Example:

Router(config)# voice register pool 4

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.

  • pool-tag —Unique sequence number of the SIP phone to be configured. Range is version and platform-dependent; type ? to display range. You can modify the upper limit for this argument by using the the max-pool command.

Step 8

dialplan dialplan-tag

Example:

Router(config-register-pool)# dialplan 1

Assigns a dial plan to a SIP phone.

  • dialplan-tag —Number that identifies the dial plan to use for this SIP phone. This is the number that was used with the voice register dialplan command in Step 3. Range: 1 to 24.

Step 9

end

Example:

Router(config-register-global)# end

Exits to privileged EXEC mode.

Examples

The following example shows the configuration for dial plan 1, which is assigned to SIP phone 1: 


  voice register dialplan  1 
		 type 7940-7960-others 
		 pattern 1 2... timeout 10 user ip 
		 pattern 2 1234 user ip button 4 
		 pattern 3 65... 
		 pattern 4 1...! 
		! 
		voice register pool  1 
		 id mac 0016.9DEF.1A70 
		 type 7961GE 
		 number 1 dn 1 
		 number 2 dn 2 
		 dialplan 1 
		 dtmf-relay rtp-nte 
		 codec g711ulaw

Troubleshooting Tips for Configuring Dial Plans for SIP

If you create a dial plan by downloading a custom XML dial pattern file to flash and using the filename command, and the XML file contains an error, the dial plan might not work properly on a phone. We recommend creating a dial pattern file using the pattern command.

To remove a dial plan that was created using a custom XML file with the filename command, you must remove the dial plan from the phone, create a new configuration profile, and then use the reset command to reboot the phone. You can use the restart command after removing a dial plan from a phone only if the dial plan was created using the pattern command.

To use KPML if a matching dial plan is not found, when both a dial plan and KPML are enabled on a phone, you must configure a dial pattern with a single wildcard character (.) as the last pattern in the dial plan. For example: 

voice register dialplan 10
 type 7940-7960-others
 pattern 1 66...
 pattern 2 91.......

What to Do Next

If you are done modifying parameters for SIP phones, you must generate a new configuration profile and restart the phones. See Configuration Files for Phones.

Verify SIP Dial Plan Configuration

Procedure

Step 1

show voice register dialplan tag

This command displays the configuration information for a specific SIP dial plan.

Example:

Router# show voice register dialplan 1 

Dialplan Tag 1
Config:
  Type is 7940-7960-others
  Pattern 1 is 2..., timeout is 10, user option is ip, button is default
  Pattern 2 is 1234, timeout is 0, user option is ip, button is 4
  Pattern 3 is 65..., timeout is 0, user option is phone, button is default
  Pattern 4 is 1..., timeout is 0, user option is phone, button is default

Step 2

show voice register pool tag

This command displays the dial plan assigned to a specific SIP phone.

Example:

Router# show voice register pool 29 

Pool Tag 29
Config:
  Mac address is 0012.7F54.EDC6
  Number list 1 : DN 29
  Proxy Ip address is 0.0.0.0
  DTMF Relay is disabled
  Call Waiting is enabled
  DnD is disabled
  keep-conference is enabled
  dialplan tag is 1
  kpml signal is enabled
  service-control mechanism is not supported
.
.
.

Step 3

show voice register template tag

This command displays the dial plan assigned to a specific template.

Example:


Router# show voice register template 3 

Temp Tag 3
Config:
  Attended Transfer is disabled
  Blind Transfer is enabled
  Semi-attended Transfer is enabled
  Conference is enabled
  Caller-ID block is disabled
  DnD control is enabled
  Anonymous call block is disabled
  Voicemail is 62000, timeout 15
  Dialplan Tag is 1
  Transport type is tcp

Enable KPML on a SIP Phone

To enable KPML digit collection on a SIP phone, perform the following steps.


Restriction

  • This feature is supported only on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.

  • A dial plan assigned to a phone has priority over KPML.

Before you begin

Cisco Unified CME 4.1 or a later version.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice register pool pool-tag
  4. digit collect kpml
  5. end
  6. show voice register dial-peers

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

voice register pool pool-tag

Example:

Router(config)# voice register pool 4

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.

  • pool-tag —Unique sequence number of the SIP phone to be configured. Range is version and platform-dependent; type ? to display range. You can modify the upper limit for this argument by using the max-pool command.

Step 4

digit collect kpml

Example:

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

Enables KPML digit collection for the SIP phone.

Note

 

This command is enabled by default for supported phones in Cisco Unified CME.

Step 5

end

Example:

Router(config-register-pool)# end

Exits to privileged EXEC mode.

Step 6

show voice register dial-peers

Example:

Router# show voice register dial-peers

Displays details of all dynamically created VoIP dial peers associated with the Cisco Unified CME SIP register, including the defined digit collection method.

What to do next

If you are done modifying parameters for SIP phones, you must generate a new configuration profile and restart the phones. See Configuration Files for Phones.

Select Session-Transport Protocol for a SIP Phone

To change the session-transport protocol for a SIP phone from the default of UDP to TCP, perform the following steps.


Restriction

  • TCP is not supported as a session-transport protocol for the Cisco Unified IP Phone 7905, 7912, 7940, or 7960. If TCP is assigned to an unsupported phone, calls to that phone will not complete successfully. However, the phone can originate calls using UDP, although TCP has been assigned.

Before you begin

  • Cisco Unified CME 4.1 or a later version.

  • Directory number must be assigned to SIP phone to which configuration is to be applied. For configuration information, see Assign Directory Numbers to SIP Phones.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice register pool pool-tag
  4. session-transport { tcp | udp}
  5. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

voice register pool pool-tag

Example:

Router(config)# voice register pool 3

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone in Cisco Unified CME.

Step 4

session-transport { tcp | udp}

Example:


   Router(config-register-pool)# session-transport tcp

(Optional) Specifies the transport layer protocol that a SIP phone uses to connect to Cisco Unified CME.

  • This command can also be configured in voice register template configuration mode and applied to one or more phones. The voice register pool configuration has priority over the voice register template configuration.

Step 5

end

Example:

Router(config-register-pool)# end

Exits voice register pool configuration mode and enters privileged EXEC mode.

What to do next


Note
When TCP is used as session-transport for the SIP phones, and if the TCP Connection aging timer is less than the SIP Register expire timer; then after every TCP connection aging timer expires, the phone will be reset and will re-register to CME. If this is not desired, then modify the TCP Connection aging timer and/or SIP Register expire timer so that SIP Register expire timer is less than TCP Connection aging timer.

Disable SIP Proxy Registration for a Directory Number

To prevent a particular directory number from registering with an external SIP proxy server, perform the following steps.


Restriction
Phone numbers that are registered under a voice register dn must belong to a SIP phone that is registered in Cisco Unified CME.

Before you begin

  • Cisco Unified CME 3.4 or a later version.

  • Bulk registration is configured at system level. For configuration information, see Configure Bulk Registration.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. voice register dn dn-tag
  4. number number
  5. no-reg
  6. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

voice register dn dn-tag

Example:

Router(config-register-global)# voice register dn 1

Enters voice register dn configuration mode to define a directory number for a SIP phone, intercom line, voice port, or an MWI.

Step 4

number number

Example:

Router(config-register-dn)# number 4085550152

Defines a valid number for a directory number to be assigned to a SIP phone in Cisco Unified CME.

Step 5

no-reg

Example:

Router(config-register-dn)# no-reg

Prevents directory number being configured from registering with an external proxy server.

Step 6

end

Example:

Router(config-register-dn)# end

Exits voice register dn configuration mode and enters privileged EXEC mode.

What to do next

Modify the Global Codec

To change the global codec from the default (G.711ulaw) to G.722-64K for all calls through Cisco Unified CME, perform the following steps.


Restriction
If G.722-64K codec is configured globally and a phone does not support the codec, the fallback codec is G.711ulaw.

Before you begin

Cisco Unified CME 4.3 or later versions.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. telephony-service
  4. codec { g711-ulaw | g722-64k}
  5. service phone g722CodecSupport { 0 | 1 | 2}
  6. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

telephony-service

Example:

Router(config)# telephony-service

Enters telephony service configuration mode to set parameters for SCCP and SIP phones in Cisco Unified CME.

Step 4

codec { g711-ulaw | g722-64k}

Example:

Router(config-telephony)# codec g722-64k

Specifies the preferred codec for phones in Cisco Unified CME.

  • Required only if you want to modify codec from the default (G.711ulaw) to G.722-64K.

Step 5

service phone g722CodecSupport { 0 | 1 | 2}

Example:

Router(config)# service phone g722CodecSupport 2

Causes all phones to advertise the G.722-64K codec to Cisco Unified CME.

  • Required only if you configured the codec g722-64k command in telephony-service configuration mode.

  • g722CodecSupport —Default: 0, phone default set by manufacturer and equal to enabled or disabled.

  • Cisco phone firmware 8.2.1 or a later version is required to support the G.722-64K codec on G.722-capable SCCP phones.

  • Cisco phone firmware 8.3.1 or a later version is required to support the G.722-64K codec on G.722-capable SIP phones.

  • For SCCP only: This command can also be configured in ephone- template configuration mode and applied to one or more SCCP phones.

Step 6

end

Example:

Router(config-telephony)# end

Exits the telephony service configuration mode and enters privileged EXEC mode.

What to do next

Configure Codecs of Individual Phones for Calls Between Local Phones

To designate a codec for individual phones to ensure connectivity between a variety of phones connected to the same Cisco Unified CME router, perform the following steps for each SCCP or SIP phone.


Note
If codec values for the dial peers of an internal connection do not match, the call fails. 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. Cisco Unified CME plays no part in the negotiation.

Restriction

  • Not all phones support all codecs. To verify whether your phone supports a particular codec, see your phone documentation.

  • For SIP and SCCP phones in Cisco Unified CME: Modify the configuration for either SIP or SCCP phones to ensure that the codec for all phones match. Do not modify the configuration for both SIP and SCCP phones.

  • If G.729 is the desired codec for Cisco ATA-186 and Cisco ATA-188, then only one port of the Cisco ATA device should be configured in Cisco Unified CME. If a call is placed to the second port of the Cisco ATA device, it will be disconnected gracefully. If you want to use both Cisco ATA ports simultaneously, then configure G.711 in Cisco Unified CME.

  • If G.722-64K or iLBC codecs are configured in ephone configuration mode and the phone does not support the codec, the fallback is the global codec or G.711ulaw if the global codec is not supported. To configure a global codec, see Modify the Global Codec.

Before you begin

  • For SIP phones in Cisco Unified CME: Cisco Unified CME 3.4 or a later version.

  • For G.722-64K and iLBC codecs: Cisco Unified CME 4.3 or a later version.

  • To support G.722-64K on an individual phone: Cisco phone firmware 8.2.1 or a later version for SCCP phones and 8.3.1 or a later version for SIP phones. For information about upgrading Cisco phone firmware, see Install Cisco Unified CME Software .

  • To support iLBC on an individual phone: Cisco phone firmware 8.3.1 or a later version for SCCP and SIP phones. For information about upgrading Cisco phone firmware, see Install Cisco Unified CME Software.

  • Cisco Unified IP phone to which the codec is to be applied must be already configured. For configuration information for SIP phones, see Assign Directory Numbers to SIP Phones. For configuration information for SCCP phones, see Assign Directory Numbers to SCCP Phones.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone ephone-tag or voice register pool pool-tag
  4. codec codec-type
  5. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

ephone ephone-tag or voice register pool pool-tag

Example:

Router(config)# voice register pool 1

Enters ephone configuration mode to set phone-specific parameters for a SCCP phone in Cisco Unified CME.

or

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone in Cisco Unified CME.

Step 4

codec codec-type

Example:


Router(config-ephone)# codec g729r8 
or 
Router(config-register-pool)# codec g711alaw

Specifies the codec for the dial peer for the IP phone being configured.

  • codec-type —Type? for a list of codecs.

  • This command overrides any previously configured codec selection set with the voice-class codec command.

  • This command overrides any previously configured codec selection set with the codec command in telephony-service configuration mode.

  • SCCP only—This command can also be configured in ephone-template configuration mode and applied to one or more phones.

Step 5

end

Example:

Router(config-ephone)# end

or 

Router(config-register-pool)# end

Exits the configuration mode and enters privileged EXEC mode.

What to do next

Configure Phones for a Key System

Creating Directory Numbers for a Simple Key System on SCCP Phone

To create a set of directory numbers with the same number to be associated with multiple line buttons on an IP phone and provide support for call waiting and call transfer on a key system phone, perform the following steps.


Restriction

  • Do not configure directory numbers for a key system for dual-line mode because this does not conform to the key system one-call-per-line button usage model for which the phone is designed.

  • Provisioning support for the Cisco Unified IP Phone 7931 is available only in Cisco Unified CME 4.0(2) and later versions.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone-dn dn-tag
  4. number number [ secondary number] [ no-reg [ both | primary] ]
  5. preference preference-order
  6. no huntstop or huntstop
  7. mwi-type { visual | audio | both}
  8. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

ephone-dn dn-tag

Example:

Router(config)# ephone-dn 11

Enters ephone-dn configuration mode to create a directory number.

Step 4

number number [ secondary number] [ no-reg [ both | primary] ]

Example:

Router(config-ephone-dn)# number 101

Configures a valid phone or extension number for this directory number.

Step 5

preference preference-order

Example:

Router(config-ephone-dn)# preference 1

Sets dial-peer preference order for a directory number associated with a Cisco Unified IP phone.

  • Default: 0.

  • Increments the preference order for all subsequent instances within a set of ephone dns with the same number to be associated with a key system phone. That is, the first instance of the directory number is preference 0 by default and you must specify 1 for the second instance of the same number, 2 for the next, and so on. This allows you to create multiple buttons with the same number on an IP phone.

  • Required to support call waiting and call transfer on a key system phone.

Step 6

no huntstop or huntstop

Example:

Router(config-ephone-dn)# no huntstop

or 

Router(config-ephone-dn)# huntstop

Explicitly enables call hunting behavior for a directory number.

  • Configure no huntstop for all instances, except the final instance, within a set of ephone dns with the same number to be associated with a key system phone.

  • Required to allow call hunting across multiple line buttons with the same number on an IP phone.

or

Disables call hunting behavior for a directory number.

  • Configure the huntstop command for the final instance within a set of ephone dns with the same number to be associated with a key system phone.

  • Required to limit the call hunting to a set of multiple line buttons with the same number on an IP phone.

Step 7

mwi-type { visual | audio | both}

Example:

Router(config-ephone-dn)# mwi-type audible

Specifies the type of MWI notification to be received.

  • This command is supported only by Cisco Unified IP Phone 7931s and Cisco Unified IP Phone 7911s.

  • This command can also be configured in ephone-dn-template configuration mode. The value set in ephone-dn configuration mode has priority over the value set in ephone-dn-template mode.

Step 8

end

Example:

Router(config-ephone-dn)# end

Exits to privileged EXEC mode.

What to do next

The following example shows the configuration for six instances of directory number 101, assigned to the first six buttons of an IP phone: 


    ephone-dn 10 
			  number 101 
			  no huntstop 
			 
			 ephone-dn 11 
			  number 101 
			  preference 1 
			  no huntstop 
			 
			 ephone-dn 12 
			  number 101 
			  preference 2
			  no huntstop
			  
			 ephone-dn 13 
			  number 101
			  preference 3
			  no huntstop 
			 
			 ephone-dn 14 
			  number 101
			  preference 4 
			  no huntstop 
			 
			 ephone-dn 15 
			  number 101 
			  preference 5
			 
			 ephone 1 
			  mac-address 0001.2345.6789> 
			  type 7931 
			  button 1:10 2:11 3:12 4:13 5:14 6:15

Configure Trunk Lines for a Key System on SCCP Phone

To set up trunk lines for your key system, perform only one of the following procedures:

Configure a Simple Key System Phone Trunk Line Configuration on SCCP Phone

Perform the steps in this section to:

  • Create directory numbers corresponding to each FXO line that allows phones to have shared or private lines connected directly to the PSTN.

  • Enable direct status monitoring of the FXO port on the line button of the IP phone. The line button indicator, either a lamp or an icon depending on the phone, shows the in-use status of the FXO port during the duration of the call.


Restriction

  • Directory number with a trunk line cannot be configured for call forward, busy, or no answer.

  • Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed on IP phones.

  • Numbers entered after trunk line is seized will not appear in call history or call detail records (CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.

  • FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and NewCall softkeys.

  • FXO trunk lines do not support conference initiator dropoff.

  • FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk line before pressing the Redial button.

  • FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP phone connected. The conference initiator is unable to participate in the conference, but can place calls on other lines.

  • FXO trunk lines do not support bulk speed dial.

  • FXO port monitoring has the following restrictions:

    • Not supported before Cisco Unified CME 4.0.

    • Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports. FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.

    • Not supported for analog ports on the Cisco VG224 or Cisco ATA 180 Series.

    • T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot into a ds0-group).

  • Transfer recall and transfer-to button optimization are supported on dual-line directory numbers only in Cisco Unified CME 4.0 and later versions.

  • Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on hold, or call pickup at alert.

Before you begin

  • FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection must be configured; for example: 

    
voice-port 1/0/0 
 connection p lar-opx 801 <<----Private number

  • Dial peers for FXO port must be configured; for example: 

    
dial-peer voice 111 pots
 destination-pattern 811 <<----Trunk-tag
 port 1/0/0

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone-dn dn-tag
  4. number number [ secondary number] [ no-reg [ both | primary] ]
  5. trunk trunk-tag [ timeout seconds] monitor-port port
  6. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:
Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:
Router# configure terminal

Enters global configuration mode.

Step 3

ephone-dn dn-tag

Example:
Router(config)# ephone-dn 51

Enters ephone-dn configuration mode to create a directory number.

  • Configure this command in the default single line mode, without the dual-line keyword, when configuring a simple key system trunk line.

Step 4

number number [ secondary number] [ no-reg [ both | primary] ]

Example:
Router(config-ephone-dn)# number 801

Configures a valid phone or extension number for this directory number.

Step 5

trunk trunk-tag [ timeout seconds] monitor-port port

Example:
Router(config-ephone-dn)# trunk 811 monitor-port 1/0/0

Associates a directory number with an FXO port.

  • The monitor-port keyword is not supported before Cisco Unified CME 4.0.

  • The monitor-port keyword is not supported on directory numbers for analog ports on the Cisco VG224 or Cisco ATA 180 Series.

Step 6

end

Example:
Router(config-ephone-dn)# end

Returns to privileged EXEC mode.

Examples

The following example shows the configuration for six instances of directory number 101, assigned to the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10: 


				ephone-dn 10
			  number 101  
			  no huntstop 
			
			 ephone-dn 11 
			  number 101 
			  preference 1 
			  no huntstop 
			 
			 ephone-dn 12 
			  number 101
			  preference 2 
			  no huntstop 
			 
			 ephone-dn 13
			  number 101 
			  preference 3 
			  no huntstop
			  
			 ephone-dn 14
			  number 101
			  preference 4 
			  no huntstop
			 
			 ephone-dn 15 
			  number 101
			  preference 5
			 
			 ephone-dn 51 
			  number 801 
			  trunk 811 monitor-port 1/0/0> 
			 
			 ephone-dn 52
			  number 802 
			  trunk 812 monitor-port 1/0/1 
			 
			 ephone-dn 53 
			  number 803 
			  trunk 813 monitor-port 1/0/2
			 
			 ephone-dn 54
			  number 804 
			  trunk 814 monitor-port 1/0/3 
			
			 ephone 1  
			  mac-address 0001.2345.6789 
			  type 7931
			  button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54 
			
			 voice-port 1/0/0
			  connection plar opx 801
			 
			 voice-port 1/0/1
			  connection plar opx 802
			 
			 voice-port 1/0/2 
			  connection plar opx 803 
			  
			 voice-port 1/0/3
			  connection plar opx 804 
			 
			 dial-peer voice 811 pots 
			  destination-pattern 811 
			  port 1/0/0 
			 
			 dial-peer voice 812 pots 
			  destination-pattern 812 
			  port 1/0/1 
			 
			 dial-peer voice 813 pots
			  destination-pattern 813
			  port 1/0/2
			  
			 dial-peer voice 814 pots 
			  destination-pattern 814 
			  port 1/0/3
What to do next

You are ready to configure each individual phone and assign button numbers, line characteristics, and directory numbers to buttons on the phone. See Configure Individual IP Phones for Key System on SCCP Phone.

Configure an Advanced Key System Phone Trunk Line Configuration on SCCP Phone

Perform the steps in this section to:

  • Create directory numbers corresponding to each FXO line that allows phones to have shared or private lines connected directly to the PSTN.

  • Enable direct status monitoring of the FXO port on the line button of the IP phone. The line button indicator, either a lamp or an icon depending on the phone, shows the in-use status of the FXO port during the duration of the call.

  • Allow transferred PSTN FXO line calls to be automatically recalled if the transfer target does not answer after the specified number of seconds. The call is withdrawn from the transfer-to phone and the call resumes ringing on the phone that initiated the transfer.


Restriction

  • Ephone-dn with a trunk line cannot be configured for call forward, busy, or no answer.

  • Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed on IP phones.

  • Numbers entered after a trunk line is seized will not appear in call history or call detail records (CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.

  • FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and NewCall softkeys.

  • FXO trunk lines do not support conference initiator dropoff.

  • FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk line before pressing the Redial button.

  • FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP phone connected. The conference initiator is unable to participate in the conference, but can place calls on other lines.

  • FXO trunk lines do not support bulk speed dial.

  • FXO port monitoring has the following restrictions:

    • Not supported before Cisco Unified CME 4.0.

    • Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports. FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.

    • Not supported for analog ports on the Cisco VG224 or Cisco ATA 180 Series.

    • T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot into a ds0-group).

  • Transfer recall and transfer-to button optimization is supported on dual-line directory numbers only in Cisco Unified CME 4.0 and later.

  • Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on hold, or call pickup at alert.

  • Transfer recall is not supported for analog ports on the Cisco VG224 or Cisco ATA 180 Series.

Before you begin
  • FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection must be configured; for example: 
    voice-port 1/0/0
 connection plar-opx 801 <<----Private number
  • Dial peers for FXO port must be configured; for example: 
    dial-peer voice 111 pots
 destination-pattern 811 <<----Trunk-tag
 port 1/0/0

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone-dn dn-tag dual-line
  4. number number [ secondary number] [ no-reg [ both | primary] ]
  5. trunk digit-string [ timeout seconds] [ transfer-timeout seconds] [ monitor-port port]
  6. huntstop [ channel]
  7. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:
Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:
Router# configure terminal

Enters global configuration mode.

Step 3

ephone-dn dn-tag dual-line

Example:
Router(config)# ephone-dn 51 dual-line

Enters ephone-dn configuration mode for the purpose of creating and configuring a telephone or extension number.

  • dual-line —Required when configuring an advanced key system phone trunk line. Dual-line mode provides a second call channel for the directory number on which to place an outbound consultation call during the call transfer attempt. This also allows the phone to remain part of the call to monitor the progress of the transfer attempt and if the transfer is not answered, to pull the call back to the phone on the original PSTN line button.

Step 4

number number [ secondary number] [ no-reg [ both | primary] ]

Example:
Router(config-ephone-dn)# number 801

Configures a valid telephone number or extension number for this directory number.

Step 5

trunk digit-string [ timeout seconds] [ transfer-timeout seconds] [ monitor-port port]

Example:
Router(config-ephone-dn)# trunk 811 transfer-timeout 30 monitor-port 1/0/0

Associates this directory number with an FXO port.

  • transfer-timeout seconds —For dual-line ephone-dns only. Range: 5 to 60000. Default: Disabled.

  • The monitor-port keyword is not supported before Cisco Unified CME 4.0.

  • The monitor-port and transfer-timeout keywords are not supported on directory numbers for analog ports on the Cisco VG224 or Cisco ATA 180 Series.

Step 6

huntstop [ channel]

Example:
Router(config-ephone-dn)# huntstop channel

Disables call hunting to the second channel of this directory number if the first channel is busy or does not answer.

  • channel —Required when configuring an advanced key system phone trunk line. Reserves the second channel created by configuring dual-line mode for the ephone-dn command so that an outbound consultation call can be placed during a call transfer attempt.

Step 7

end

Example:
Router(config-ephone-dn)# end

Exits to privileged EXEC mode.

Examples

The following example shows the configuration for six instances of directory number 101, assigned to the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10. These four PSTN line appearances are configured as dual lines to provide a second call channel on which to place an outbound consultation call during a call transfer attempt. This configuration allows the phone to remain part of the call to monitor the progress of the transfer attempt, and if the transfer is not answered, to pull the call back to the phone on the original PSTN line button. 


    ephone-dn 10 
			  number 101  
			  no huntstop
			
			 ephone-dn 11 
			  number 101 
			  preference 1 
			  no huntstop 
			  
			 ephone-dn 12 
			  number 101 
			  preference 2 
			  no huntstop 
			   
			 ephone-dn 13 
			  number 101 
			  preference 3
			  no huntstop 
			  
			 ephone-dn 14 
			  number 101 
			  preference 4 
			  no huntstop
			  
			 ephone-dn 15 
			  number 101
			  preference 5 
			 
			 ephone-dn 51 dual-line 
			  number 801 
			  trunk 811 transfer-timeout 30 monitor-port 1/0/0 
			  huntstop channel 
			 
			 ephone-dn 52 dual-line
			  number 802 
			  trunk 812 transfer-timeout 30 monitor-port 1/0/1 
			  huntstop channel 
			  
			 ephone-dn 53 dual-line
			  number 803 
			  trunk 813 transfer-timeout 30 monitor-port 1/0/2 
			  huntstop channel 
			  
			 ephone-dn 54 dual-line
			  number 804> 
			  trunk 814 transfer-timeout 30 monitor-port 1/0/3 
			  huntstop channel 
			  
			 ephone 1  
			  mac-address 0001.2345.6789 
			  type 7931 
			  button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54 
			  
			 voice-port 1/0/0 
			  connection plar opx 801 
			  
			 voice-port 1/0/1 
			  connection plar opx 802 
			 
			 voice-port 1/0/2 
			  connection plar opx 803 
			  
			 voice-port 1/0/3
			  connection plar opx 804 
			  
			 dial-peer voice 811 pots 
			  destination-pattern 811 
			  port 1/0/0 
			  
			 dial-peer voice 812 pots 
			  destination-pattern 812 
			  port 1/0/1 
			 
			 dial-peer voice 813 pots 
			  destination-pattern 813 
			  port 1/0/2 
			  
			 dial-peer voice 814 pots 
			  destination-pattern 814 
			  port 1/0/3

Configure Individual IP Phones for Key System on SCCP Phone

To assign button numbers, line characteristics, and directory numbers to buttons on an individual phone that will operate as a key system phone, perform the following steps.


Restriction

  • Provisioning for Cisco Unified IP Phone 7931G is available only in Cisco Unified CME 4.0(2) and later versions.

  • Cisco Unified IP Phone 7931G can support only one call waiting overlaid per directory number.

  • Cisco Unified IP Phone 7931G cannot support overlays that contain directory numbers configured for dual-line mode.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ephone phone-tag
  4. mac-address [ mac-address]
  5. type phone-type
  6. button button-number { separator} dn-tag [ ,dn-tag...] [ button-number{ x} overlay-button-number] [ button-number...]
  7. mwi-line line-number
  8. end

DETAILED STEPS

  Command or Action Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

ephone phone-tag

Example:

Router(config)# ephone 1

Enters ephone configuration mode.

Step 4

mac-address [ mac-address]

Example:

Router(config-ephone)# mac-address 0001.2345.6789

Specifies the MAC address of the IP phone that is being configured.

Step 5

type phone-type

Example:

Router(config-ephone)# type 7931

Specifies the type of phone that is being configured.

Step 6

button button-number { separator} dn-tag [ ,dn-tag...] [ button-number{ x} overlay-button-number] [ button-number...]

Example:

Router(config-ephone)# button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54

Associates a button number and line characteristics with an ephone-dn. Maximum number of buttons is determined by phone type.

Tip

 

The line button layout for the Cisco Unified IP Phone 7931G is a bottom-up array. Button 1 is at the bottom right of the array and button 24 is at the top left of the array.

Step 7

mwi-line line-number

Example:

Router(config-ephone)# mwi-line 3

Selects a phone line to receive MWI treatment; when a message is waiting for the selected line, the message waiting indicator is activated.

  • line-number —Range: 1 to 34. Default: 1.

Step 8

end

Example:

Router(config-ephone)# end

Exits ephone configuration mode and enters privileged EXEC mode.

What to do next

Configure Cisco ATA, Analog Phone Support, Remote Phones, Cisco IP Communicator, and Secure IP Phone (IP-STE)

Configure Cisco ATA Support in SCCP Mode

To enable an analog phone that uses a Cisco ATA to register with Cisco Unified CME, perform the following steps.


Restriction
For a Cisco ATA that is registered to a Cisco Unified CME system to participate in fax calls, it must have its ConnectMode parameter set to use the same RTP payload type as the Cisco voice gateway that is performing the fax pass-through. Cisco voice gateways use standard payload type 0/8, which is selected on Cisco ATAs by setting bit 2 of the ConnectMode parameter to 1. For more information, see the Parameters and Defaults chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).

Procedure

Step 1

Install the Cisco ATA.

See the Installing the Cisco ATA chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).

Step 2

Configure the Cisco ATA.

See the Configuring the Cisco ATA for SCCP chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).

Step 3

Upgrade the firmware to the latest Cisco ATA image.

If you are using either the v2.14 or v2.14ms Cisco ATA 186 image based on the 2.14 020315a build for H.323/SIP or the 2.14 020415a build for MGCP or SCCP, you must upgrade to the latest version to install a security patch. This patch fixes a security hole in the Cisco ATA Web server that allows users to bypass the user interface password.

For information about upgrading firmware, see Install Cisco Unified CME Software. Alternatively, you can use a manual method, as described in the Upgrading the Cisco ATA Signaling Image chapter of Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP (version 3.0).

Step 4

Set the following network parameters on the Cisco ATA:

  • DHCP parameter to 1 (enabled).

  • TFTP parameter to 1 (enabled).

  • TFTPURL parameter to the IP address of the router running Cisco Unified CME.

  • SID0 parameter to a period (. ) or the MAC address of the Cisco ATA (to enable the first port).

  • SID1 parameter to a period (. ) or a modified version the Cisco ATA’s MAC address, with the first two hexadecimal numbers removed and 01 appended to the end, if you want to use the second port. For example, if the MAC address of the Cisco ATA is 00012D01073D, set SID1 to 012D01073D01.

  • Nprintf parameter to the IP address and port number of the host to which all Cisco ATA debug messages are sent. The port number is usually set to 9001.

  • To prevent tampering and unauthorized access to the Cisco ATA 186, you can disable the web-based configuration. However, if you disable the web configuration page, you must use either a TFTP server or the voice configuration menu to configure the Cisco ATA 186.

Step 5

In Cisco Unified CME, configure analog phones that use a Cisco ATA in the same way as a Cisco Unified IP phone. In the type command, use the ata keyword. For information on how to provision phones, see Create Directory Numbers for SCCP Phones.

What to do next

Configure Cisco ATA Support in SIP Mode

Cisco ATA 187, 190, and 191 support SIP mode. To enable an analog phone that uses a Cisco ATA 191 to register with Unified CME, perform the following steps.


Restriction

  • For a Cisco ATA that is registered to a Unified CME system to participate in fax calls, it must have its ConnectMode parameter set to use the same RTP payload type as the Cisco voice gateway that is performing the Fax pass-through. Cisco voice gateways use standard payload type 0/8, which is selected on Cisco ATAs by setting bit 2 of the ConnectMode parameter to 1. For more information, see the Configure Fax Services chapter in Cisco ATA 191 Analog Telephone Adapter Administration Guide for Cisco Unified Communications Manager.

  • If both ports of a Cisco ATA 191 are configured as shared line, then a call put on hold on one port cannot be resumed at the other port.

Procedure

Step 1

Install the Cisco ATA.

See the Install the ATA 191 chapter in Cisco ATA 191 Analog Telephone Adapter Administration Guide for Cisco Unified Communications Manager.

Step 2

Configure the Cisco ATA.

See the Configure the ATA 191 chapter in Cisco ATA 191 Analog Telephone Adapter Administration Guide for Cisco Unified Communications Manager.

Step 3

Upgrade the firmware to the latest Cisco ATA image. For more information, see Configure Firmware Upgrade for ATA in SIP Mode.

Step 4

In Cisco Unified CME, configure analog phones that use a Cisco ATA in the same way as a Cisco Unified IP phone. In the type command that is configured under voice register pool configuration mode, use the ATA-191 keyword. For information on how to provision phones, see Create Directory Numbers for SIP Phones.

Configure Firmware Upgrade for ATA in SIP Mode

Cisco ATA 187, 190, and 191 support SIP mode. To configure firmware upgrade for ATA 190 in SIP mode with Unified CME, perform the following steps.

You can specify the Cisco ATA 191 phone type using the CLI command type as shown: 

Router(config)# voice register pool 1
Router(config-register-pool)# type ATA-191
Procedure

Step 1

Copy the firmware files to router flash memory.

For example, ATA190.1-1-2-005.loads and ATA190.1-1-2-005.bin.sgn are firmware files for ATA 190.

The firmware file for ATA 12.0(1) tat is supported in Unified CME is cmterm-ata191.12-0-1SR1-1.zip.

Step 2

Create TFTP bindings for the firmware files. 

 Router(confg)#tftp-server Flash:ATA190.1-1-2-005.bin.sgn
Router(confg) tftp-server Flash:ATA190.1-1-2-005.loads

Step 3

Specify the load using loads command under voice register global configuration mode. 

 Router(confg)#voice register global
Router(confg-register-global) load ATA-190 ATA190.1-1-2-005

Step 4

Configure a pool for ATA phone to be upgraded.

Step 5

Create CNF files using create profile CLI command under voice register global configuration mode.

Step 6

Restart the ATA by unplugging and re-plugging or by executing the reset command.

Cisco ATA 190/191 takes around 5 mins to upgrade the firmware.

Verify the new firmware using show voice register pool phone-load CLI command
 
Router#show voice register pool phone-load
Pool Device Name      Current-Version         Previous-Version
==== ===========      ================        ================
1    SEP34DB²D18001C  Cisco/ATA190-1.1.2(005) Cisco/ATA190-1.1.1(003)

Verify Cisco ATA Support

Use the show ephone ata command to display SCCP phone configurations with the type ata command.

The following is sample output for a Cisco Unified CME configured for two analog phones using a Cisco ATA with MAC address 000F.F758.E70E: 

 
	 ephone-30 Mac:000F.F758.E70E TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 1 and Server in ver 1
	 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
	 IP:1.4.188.72 15325 ATA Phone  keepalive 7 max_line 2 dual-line
	 button 1: dn 80 number 8080 CH1   IDLE         CH2 IDLE 
	 
	 ephone-31 Mac:0FF7.58E7.0E01 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 1 and Server in ver 1
	 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:3
	 IP:1.4.188.72 15400 ATA Phone  keepalive 7 max_line 2 dual-line
	 button 1: dn 81 number 8081 CH1   IDLE         CH2 IDLE

Troubleshooting Cisco ATA Support

Use the debug ephone detail command to diagnose problems with analog phones that use Cisco ATAs.

Call Pickup and Group Call Pickup with Cisco ATA

Most of the procedures for using Cisco ATAs with Cisco Unified CME are the same as those for using Cisco ATAs with Cisco Unified Communications Manager, as described in the How to Use Pre-Call and Mid-Call Services chapter of Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SCCP (version 3.0). However, the call pickup and group call pickup procedures are different when using Cisco ATAs with Cisco Unified CME, as described below:

Call Pickup

When using Cisco ATAs with Cisco Unified CME:

  • To pickup the last parked call, press **3* .

  • To pickup a call on a specific extension, press **3 and enter the extension number.

  • To pickup a call from a park slot, press **3 and enter the park slot number.

Group Call Pickup

When using Cisco ATAs with Cisco Unified CME:

  • To answer a phone within your call pickup group, press **4* .

  • To answer a phone outside of your call pickup group, press **4 and the group ID number.


Note