Cisco Unified Communications Manager Express System Administrator Guide
Integrating Voice Mail
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Table of Contents

Integrating Voice Mail

Contents

Prerequisites

Information About Voice-Mail Integration

CiscoUnityConnection Integration

CiscoUnityExpress Integration

CiscoUnity Integration

DTMF Integration for Legacy Voice-Mail Applications

Mailbox Selection Policy

RFC 2833 DTMF MTP Passthrough

MWI Line Selection

AMWI

SIP MWI Prefix Specification

SIP MWI - QSIG Translation

VMWI

Transfer to Voice Mail

Live Record

CiscoUnityExpress AXL Enhancement

How to Configure Voice-Mail Integration

SCCP: Configuring a Voice Mailbox Pilot Number

Prerequisites

What to Do Next

SCCP: Configuring a Mailbox Selection Policy

SCCP: Setting a Mailbox Selection Policy for CiscoUnityExpress or a PBX Voice-Mail Number

SCCP: Setting Mailbox Selection Policy for CiscoUnity

SCCP: Enabling Transfer to Voice Mail

Prerequisites

Restrictions

Example

What to Do Next

SCCP: Configuring Live Record

Prerequisites

Restrictions

Example

SIP: Configuring a Voice Mailbox Pilot Number

Prerequisites

What to Do Next

Enabling DTMF Integration

Enabling DTMF Integration for Analog Voice-Mail Applications

Enabling DTMF Integration Using RFC 2833

Enabling DTMF Integration Using SIP NOTIFY

SCCP: Configuring a Phone for MWI Outcall

Prerequisites

Restrictions

SIP: Enabling MWI at the System-Level

Prerequisites

SIP: Configuring a Directory Number for MWI

SIP: Defining Pilot Call Back Number for MWI Outcall

SIP: Configuring a Directory Number for MWI NOTIFY

Enabling SIP MWI Prefix Specification

Prerequisites

SIP: Configuring VMWI

Prerequisites

Verifying Voice-Mail Integration

Configuration Examples for Voice-Mail Integration

Mailbox Selection Policy for SCCP Phones: Example

Voice Mailbox for SIP Phones: Example

DTMF Integration Using RFC 2833: Example

DTMF Integration Using SIP Notify: Example

DTMF Integration for Legacy Voice-Mail Applications: Example

SCCP Phone Line for MWI: Example

SIP MWI Prefix Specification: Example

SIP Directory Number for MWI Outcall: Example

SIP Directory Number for MWI Unsolicited Notify: Example

SIP Directory Number for MWI Subscribe/NOTIFY: Example

Additional References

Related Documents

Technical Assistance

Feature Information for Voice-Mail Integration

Integrating Voice Mail

Last Updated: June 07, 2010

 

This chapter describes how to integrate your voice-mail system with Cisco Unified Communications Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a list of the versions in which each feature is supported, see the “Feature Information for Voice-Mail Integration” section.

Prerequisites

  • Calls can be successfully completed between phones on the same Cisco Unified CME router.
  • If your voice-mail system is something other than Cisco Unity Express, such as Cisco Unity, voice mail must be installed and configured on your network.
  • If your voice-mail system is Cisco Unity Express:

Note When you order Cisco Unity Express, Cisco Unity Express software and the purchased license are installed on the module at the factory. Spare modules also ship with the software and license installed. If you are adding Cisco Unity Express to an existing Cisco router, you will be required to install hardware and software components.


Interface module for Cisco Unity Express is installed. For information about the AIM-CUE or NM-CUE, access documents located at http://www.cisco.com/en/US/products/hw/modules/ps2797/prod_installation_guides_list.html

The recommended Cisco IOS release and feature set plus the necessary Cisco Unified CME phone firmware and GUI files to support Cisco Unity Express are installed on the Cisco Unified CME router.

If the GUI files are not installed, see the “Installing Cisco Unified CME Software” section .

To determine whether the Cisco IOS software release and Cisco Unified CME software version are compatible with the Cisco Unity Express version, Cisco router model, and Cisco Unity Express hardware that you are using, see the Cisco Unity Express Compatibility Matrix .

To verify installed Cisco Unity Express software version, enter the Cisco Unity Express command environment and use the show software version user EXEC command. For information about the command environment, see the appropriate Cisco Unity Express CLI Administrator Guide at http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/roadmap/cuedocs.html .

The proper license for Cisco Unified CME, not Cisco Unified Communications Manager, is installed. To verify installed license, enter the Cisco Unity Express command environment and use the show software license user EXEC command. For information about the command environment, see the appropriate Cisco Unity Express CLI Administrator Guide at http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/roadmap/cuedocs.html .

This is an example of the Cisco Unified CME license:

se-10-0-0-0> show software licenses
Core:
- application mode: CCME
- total usable system ports: 8
Voicemail/Auto Attendant:
- max system mailbox capacity time: 6000
- max general delivery mailboxes: 15
- max personal mailboxes: 50
Languages:
- max installed languages: 1
- max enabled languages: 1

Voicemail and Auto Attendant (AA) applications are configured. For configuration information, see “Configuring the System Using the Initialization Wizard” in the appropriate Cisco Unity Express GUI Administrator Guide at http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/roadmap/cuedocs.html.

Information About Voice-Mail Integration

To enable voice-mail support, you should understand the following concepts:

Cisco Unity Connection Integration

Cisco Unity Connection transparently integrates messaging and voice recognition components with your data network to provide continuous global access to calls and messages. These advanced, convergence-based communication services help you use voice commands to place calls or listen to messages in “hands-free” mode and check voice messages from your desktop, either integrated into an e-mail inbox or from a Web browser. Cisco Unity Connection also features robust automated-attendant functions that include intelligent routing and easily customizable call-screening and message-notification options.

For instructions on how to integrate Cisco Unified CME with Cisco Unity Connection, see the Cisco CallManager Express 3.x Integration Guide for Cisco Unity Connection 1.1 .

Cisco Unity Express Integration

Cisco Unity Express offers easy, one-touch access to messages and commonly used voice-mail features that enable users to reply, forward, and save messages. To improve message management, users can create alternate greetings, access envelope information, and mark or play messages based on privacy or urgency. For instructions on how to configure Cisco Unity Express, see the administrator guides for Cisco Unity Express .

For configuration information, see the “Enabling DTMF Integration Using SIP NOTIFY” section.


NoteCisco Unified CME and Cisco Unity Express must both be configured before they can be integrated. Cisco Unified CME and Cisco Unity Express must both be configured before they can be integrated.


Cisco Unity Integration

Cisco Unity is a Microsoft Windows-based communications solution that brings you voice mail and unified messaging and integrates them with the desktop applications you use daily. Cisco Unity gives you the ability to access all of your messages, voice, fax, and e-mail, by using your desktop PC, a touchtone phone, or the Internet. The Cisco Unity voice mail system supports voice-mail integration with Cisco Unified CME. This integration requires that you configure the Cisco Unified CME router and Cisco Unity software to get voice-mail service.

For configuration instructions, see the “Enabling DTMF Integration Using RFC 2833” section.

DTMF Integration for Legacy Voice-Mail Applications

For dual-tone multifrequency (DTMF) integrations, information on how to route incoming or forwarded calls is sent by a telephone system in the form of DTMF digits. The DTMF digits are sent in a pattern that is based on the integration file in the voice-mail system connected to the Cisco Unified CME router. These patterns are required for DTMF integration of Cisco Unified CME with most voice-mail systems. Voice-mail systems are designed to respond to DTMF after the system answers the incoming calls.

After configuring the DTMF integration patterns on the Cisco Unified CME router, you set up the integration files on the third-party legacy voice-mail system by following the instructions in the documents that accompany the voice-mail system. You must design the DTMF integration patterns appropriately so that the voice-mail system and the Cisco Unified CME router work with each other.

For configuration information, see the “Enabling DTMF Integration for Analog Voice-Mail Applications” section.

Mailbox Selection Policy

Typically a voice-mail system uses the number that a caller has dialed to determine the mailbox to which a call should be sent. However, if a call has been diverted several times before reaching the voice-mail system, the mailbox that is selected might vary for different types of voice-mail systems. For example, Cisco Unity Express uses the last number to which the call was diverted before it was sent to voice mail as the mailbox number. Cisco Unity and some legacy PBX systems use the originally called number as the mailbox number.

The Mailbox Selection Policy feature allows you to provision the following options from the Cisco Unified CME configuration.

  • For Cisco Unity Express, you can select the originally dialed number.
  • For PBX voice-mail systems, you can select the last number to which the call was diverted before it was sent to voice mail. This option is configured on the outgoing dial peer for the voice-mail system's pilot number.
  • For Cisco Unity voice mail, you can select the last number to which the call was diverted before it was sent to voice mail. This option is configured on the ephone-dn that is associated with the voice-mail pilot number.

To enable Mailbox Selection Policy, see the “SCCP: Setting a Mailbox Selection Policy for Cisco Unity Express or a PBX Voice-Mail Number” section or the “SCCP: Setting Mailbox Selection Policy for Cisco Unity” section.

RFC 2833 DTMF MTP Passthrough

In Cisco Unified CME 4.1, the RFC 2833 Dual-Tone Multifrequency (DTMF) Media Termination Point (MTP) Passthrough feature provides the capability to pass DTMF tones transparently between SIP endpoints that require transcoding or Resource Reservation Protocol (RSVP) agents.

This feature supports DTMF Relay across SIP WAN devices that support RFC 2833, such as Cisco Unity and SIP trunks. Devices registered to a Cisco Unified CME SIP back-to-back user agent (B2BUA) can exchange RFC 2833 DTMF MTP with other devices that are not registered with the Cisco Unified CME SIP B2BUA, or with devices that are registered in one of the following:

  • Local or remote Cisco Unified CME
  • Cisco Unified Communications Manager
  • Third party proxy

By default, the RFC 2833 DTMF MTP Passthrough feature uses payload type 101 on MTP, and MTP accepts all the other dynamic payload types if it is indicated by Cisco Unified CME. For configuration information, see the “Enabling DTMF Integration Using RFC 2833” section.

MWI Line Selection

Message waiting indicator (MWI) line selection allows you to choose the phone line that is monitored for voice-mail messages and that lights an indicator when messages are present.

Before Cisco Unified CME 4.0, the MWI lamp on a phone running SCCP could be associated only with the primary line of the phone.

In Cisco Unified CME 4.0 and later versions, you can designate a phone line other than the primary line to be associated with the MWI lamp. Lines other than the one associated with the MWI lamp display an envelope icon when a message is waiting. A logical phone “line” is not the same as a phone button. A button with one or more directory numbers is considered one line. A button with no directory number assigned does not count as a line.

In Cisco Unified CME 4.0 and later versions, a SIP directory number that is used for call forward all, presence BLF status, and MWI features must be configured by using the dn keyword in the number command; direct line numbers are not supported.

For configuration information, see the“SCCP: Configuring a Voice Mailbox Pilot Number” section or “SIP: Configuring a Directory Number for MWI” section.

AMWI

The AMWI (Audible Message Line Indicator) feature provides a special stutter dial tone to indicate message waiting. This is an accessibility feature for vision-impaired phone users. The stutter dial tone is defined as 10 ms ON, 100 ms OFF, repeat 10 times, then steady on.

In Cisco Unified CME 4.0(3), you can configure the AMWI feature on the Cisco Unified IP Phone 7911 and Cisco Unified IP Phone 7931G to receive audible, visual, or audible and visual MWI notification from an external voice-messaging system. AMWI cannot be enabled unless the number command is already configured for the IP phone to be configured.

Cisco Unified CME applies the following logic based on the capabilities of the IP phone and how MWI is configured:

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

If a phone supports (visual) MWI and AMWI and both options are configured for the phone, activate the Message Waiting light and send the stutter dial tone to the phone when it goes off-hook.

For configuration informations, see the “SCCP: Configuring a Phone for MWI Outcall” section.

SIP MWI Prefix Specification

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

For example, an MWI message might indicate that the central mailbox number 555-0123 has a voice message. In this example, the digits 555 are set as the prefix string or site identifier using the mwi prefix command. The local Cisco Unified CME system is able to convert 555-0123 to 0123 and deliver the MWI to the correct phone. Without this prefix string manipulation, the system would reject an MWI for 555-0123 as not matching the local Cisco Unified CME extension 0123.

To enable SIP MWI Prefix Specification, see the “Enabling SIP MWI Prefix Specification” section.

SIP MWI - QSIG Translation

In Cisco Unified CME 4.1 and later, the SIP MWI - QSIG Translation feature extends MWI functionality for SIP MWI and QSIG MWI interoperation to enable sending and receiving MWI over QSIG to a PBX.

When the SIP Unsolicited NOTIFY is received from voice mail, the Cisco router translates this event to activate QSIG MWI to the PBX, via PSTN. The PBX will switch on, or off, the MWI lamp on the corresponding IP phone. This feature supports only Unsolicited NOTIFY. Subscribe NOTIFY is not supported by this feature.

In Figure 1, the Cisco router receives the SIP Unsolicited NOTIFY, performs the protocol translation, and initiates the QSIG MWI call to the PBX, where it is routed to the appropriate phone.

Figure 1 SIP MWI to ISDN QSIG When Voice Mail and Cisco Router are On the Same LAN

 

It makes no difference if the SIP Unsolicited NOTIFY is received via LAN or WAN if the PBX is connected to the Cisco router, and not to the remote voice-mail server.

In Figure 2, a voice mail server and Cisco Unified CME are connected to the same LAN and a remote Cisco Unified CME is connected across the WAN. In this scenario, the protocol translation is performed at the remote Cisco router and the QSIG MWI message is sent to the PBX.

Figure 2 SIP MWI to ISDN QSIG When PBX is Connected to a Remote Cisco Router

 

VMWI

There are two types of visual message waiting indicator (VMWI) features: Frequency-shift Keying (FSK) and DC voltage. The message-waiting lamp can be enabled to flash on an analog phone that requires an FSK message to activate a visual indicator. The DC Voltage VMWI feature is used to flash the message-waiting lamp on an analog phone which requires DC voltage instead of an FSK message. For all other applications, such as MGCP, FSK VMWI is used even if the voice gateway is configured for DC voltage VMWI. The configuration for DC voltage VMWI is supported only for Foreign Exchange Station (FXS) ports on the Cisco VG224 analog voice gateway with analog device version V1.3 and V2.1.

The Cisco VG224 can only support 12 Ringer Equivalency Number (REN) for ringing 24 onboard analog FXS voice ports. To support ringing and DC Voltage VMWI for 24 analog voice ports, stagger-ringing logic is used to maximize the limited REN resource. When a system runs out of REN because too many voice ports are being rung, the MWI lamp temporarily turns off to free up REN to ring the voice ports.

DC voltage VMWI is also temporarily turned off any time the port's operational state is no longer idle and onhook, such as when one of the following events occur:

  • Incoming call on voice port
  • Phone goes off hook
  • The voice port is shut down or busied out

Once the operational state of the port changes to idle and onhook again, the MWI lamp resumes flashing until the application receives a requests to clear it; for example, if there are no more waiting messages.

For configuration information, see the “Transfer to Voice Mail” section.

Transfer to Voice Mail

The Transfer to Voice Mail feature allows a phone user to transfer a caller directly to a voice-mail extension. The user presses the TrnsfVM soft key to place the call on hold, enters the extension number, and then commits the transfer by pressing the TrnsfVM soft key again. The caller hears the complete voice mail greeting. This feature is supported using the TrnsfVM soft key or feature access code (FAC).

For example, a receptionist might screen calls for five managers. If a call comes in for a manager who is not available, the receptionist can transfer the caller to the manager's voice-mail extension by using the TrnsfVM soft key and the caller hears the personal greeting of the individual manager.

For configuration information, see the “SCCP: Enabling Transfer to Voice Mail” section.

Live Record

The Live Record feature enables IP phone users in a Cisco Unified CME system to record a phone conversation if Cisco Unity Express is the voice mail system. An audible notification, either by announcement or by periodic beep, alerts participants that the conversation is being recorded. The playing of the announcement or beep is under the control of Cisco Unity Express.

Live Record is supported for two-party calls and ad hoc conferences. In normal record mode, the conversation is recorded after the LiveRcd soft key is pressed. This puts the other party on-hold and initiates a call to Cisco Unity Express at the configured live-record number. To stop the recording session, the phone user presses the LiveRcd soft key again, which toggles between on and off.

The Live-Record number is configured globally and must match the number configured in Cisco Unity Express. You can control the availability of the feature on individual phones by modifying the display of the LiveRcd soft key using an ephone template. This feature must be enabled on both Cisco Unified CME and Cisco Unity Express.

To enable Live Record in Cisco Unified CME, see the “SCCP: Configuring Live Record” section.

Cisco Unity Express AXL Enhancement

In Cisco Unified CME 7.0(1) and later versions, the Cisco Unity Express AXL enhancement in Cisco Unified CME provides better administrative integration between Cisco Unified CME and Cisco Unity Express by automatically synchronizing passwords.

No configuration is required to enable this feature.

How to Configure Voice-Mail Integration

This section contains the following tasks:

SCCP: Configuring a Voice Mailbox Pilot Number

To configure the telephone number that is speed-dialed when the Message button on a SCCP phone is pressed, perform the following steps.


NoteThe same telephone number is configured for voice messaging for all SCCP phones in Cisco Unified CME. The same telephone number is configured for voice messaging for all SCCP phones in Cisco Unified CME.


Prerequisites

  • Voicemail phone number must be a valid number; directory number and number for voicemail phone number must be configured. For configuration information, see Configuring Phones to Make Basic Calls.

SUMMARY STEPS

1. enable

2. configure terminal

3. telephony-service

4. voicemail phone-number

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

telephony-service

 

Router(config)# telephony-service

Enters voice register global configuration mode to set parameters for all supported phones in Cisco Unified CME.

Step 4

voicemail phone-number

 

Router(config-telephony)# voice mail 0123

Defines the telephone number that is speed-dialed when the Messages button on a Cisco Unified IP phone is pressed.

  • phone-number— Same phone number is configured for voice messaging for all SCCP phones in a Cisco Unified CME.

Step 5

end

 

Router(config-telephony)# end

Exits to privileged EXEC mode.

What to Do Next

SCCP: Configuring a Mailbox Selection Policy

Perform one of the following tasks, depending on which voice-mail application is used:

SCCP: Setting a Mailbox Selection Policy for Cisco Unity Express or a PBX Voice-Mail Number

To set a policy for selecting a mailbox for calls from a Cisco Unified CME system that are diverted before being sent to a Cisco Unity Express or PBX voice-mail pilot number, perform the following steps.

Prerequisites

Cisco Unified CME 4.0 or a later version.

Restrictions

In the following scenarios, the mailbox selection policy can fail to work properly:

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

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip
or
dial-peer voice tag pots

4. mailbox-selection [ last-redirect-num | orig-called-num ]

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

dial-peer voice tag voip

or

dial-peer voice tag pots

 

Router(config)# dial-peer voice 7000 voip

or

Router(config)# dial-peer voice 35 pots

Enters dial-peer configuration mode.

  • tag —Identifies the dial peer. Valid entries are 1 to 2147483647.

Note Use this command on the outbound dial peer associated with the pilot number of the voice-mail system. For systems using Cisco Unity Express, this is a VoIP dial peer. For systems using PBX-based voice mail, this is a POTS dial peer.

Step 4

mailbox-selection [ last-redirect-num | orig-called-num ]

 

Router(config-dial-peer)# mailbox-selection orig-called-num

Sets a policy for selecting a mailbox for calls that are diverted before being sent to a voice-mail line.

  • last-redirect-num —(PBX voice mail only) The mailbox number to which the call will be sent is the last number to divert the call (the number that sends the call to the voice-mail pilot number).
  • orig-called-num —(Cisco Unity Express only) The mailbox number to which the call will be sent is the number that was originally dialed before the call was diverted.

Step 5

end

 

Router(config-ephone-dn)# end

Returns to privileged EXEC mode.

What to Do Next

SCCP: Setting Mailbox Selection Policy for Cisco Unity

To set a policy for selecting a mailbox for calls that are diverted before being sent to a Cisco Unity voice-mail pilot number, perform the following steps.

Prerequisites

  • Cisco Unified CME 4.0 or a later version.
  • Directory number to be configured is associated with a voice mailbox.

Restrictions

This feature might not work properly in certain network topologies, including when:

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

SUMMARY STEPS

1. enable

2. configure terminal

3. ephone-dn dn-tag

4. mailbox-selection last-redirect-num

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

exit

 

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 4

ephone-dn

 

Router(config)# ephone-dn 752

Enters ephone-dn configuration mode.

Step 5

mailbox-selection [ last-redirect-num ]

 

Router(config-ephone-dn)# mailbox-selection last-redirect-num

Sets a policy for selecting a mailbox for calls that are diverted before being sent to a Cisco Unity voice-mail pilot number.

Step 6

end

 

Router(config-ephone-dn)# end

Returns to privileged EXEC mode.

What to Do Next

SCCP: Enabling Transfer to Voice Mail

To enable a phone user to transfer a call to voice mail by using the TrnsfVM soft key or a FAC, perform the following steps.

Prerequisites

  • Cisco Unified CME 4.3 or a later version.
  • Cisco Unity Express 3.0 or a later version, installed and configured.
  • For information about standard and custom FACs, seeConfiguring Feature Access Codes.

Restrictions

The TrnsfVM soft key is not supported on the Cisco Unified IP Phone 7905, 7912, or 7921, or analog phones connected to the Cisco VG224 or Cisco ATA. These phones support the trnsfvm FAC.

SUMMARY STEPS

1. enable

2. configure terminal

3. ephone-template template-tag

4. softkeys connected {[ Acct ] [ ConfList ] [ Confrn ] [ Endcall ] [ Flash ] [ HLog ] [ Hold ] [ Join ] [ LiveRcd ] [ Park ] [ RmLstC ] [ Select ] [ TrnsfVM ] [ Trnsfer ]}

5. exit

6. ephone phone-tag

7. ephone-template template-tag

8. exit

9. telephony-service

10. voicemail phone-number

11. fac { standard | custom trnsfvm custom-fac }

12. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

ephone-template template-tag

 

Router(config)# ephone-template 5

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

  • template-tag —Unique identifier for the ephone template. Range: 1 to 20.

Step 4

softkeys connected {[ Acct ] [ ConfList ] [ Confrn ] [ Endcall ] [ Flash ] [ HLog ] [ Hold ] [ Join ] [ LiveRcd ] [ Park ] [ RmLstC ] [ Select ] [ TrnsfVM ] [ Trnsfer ]}

 

Router(config-ephone-template)# softkeys connected TrnsfVM Park Acct ConfList Confrn Endcall Trnsfer Hold

(Optional) Modifies the order and type of soft keys that display on an IP phone during the connected call state.

  • You can enter any of the keywords in any order.
  • Default is all soft keys are displayed in alphabetical order.
  • Any soft key that is not explicitly defined is disabled.

Step 5

exit

 

Router(config-ephone-template)# exit

Exits ephone-template configuration mode.

Step 6

ephone phone-tag

 

Router(config)# ephone 12

Enters ephone configuration mode.

  • phone-tag —Unique number that identifies this ephone during configuration tasks.

Step 7

ephone-template template-tag

 

Router(config-ephone)# ephone-template 5

Applies the ephone template to the phone.

  • template-tag —Unique identifier of the ephone template that you created in Step 3.

Step 8

exit

 

Router(config-ephone)# exit

Exits ephone configuration mode.

Step 9

telephony-service

 

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 10

voicemail phone-number

 

Router(config-telephony)# voicemail 8900

Defines the telephone number that is speed-dialed when the Messages button on a Cisco Unified IP phone is pressed.

  • phone-number— Same phone number is configured for voice messaging for all SCCP phones in a Cisco Unified CME.

Step 11

fac { standard | custom trnsfvm custom-fac }

 

Router(config-telephony)# fac custom trnsfvm #22

Enables standard FACs or creates a custom FAC or alias.

  • standard —Enables standard FACs for all phones. Standard FAC for transfer to voice mail is *6.
  • custom —Creates a custom FAC for a FAC type.
  • custom-fac —User-defined code to be dialed using the keypad on an IP or analog phone. Custom FAC can be up to 256 characters long and contain numbers 0 to 9 and * and #.

Step 12

end

 

Router(config-telephony)# end

Returns to privileged EXEC mode.

Example

The following example shows a configuration where the display order of the TrnsfVM soft key is modified for the connected call state in ephone template 5 and assigned to ephone 12. A custom FAC for transfer to voice mail is set to #22.

telephony-service

max-ephones 100

max-dn 240

timeouts transfer-recall 60

voicemail 8900

max-conferences 8 gain -6

transfer-system full-consult

fac custom trnsfvm #22

!

!

ephone-template 5

softkeys connected TrnsfVM Park Acct ConfList Confrn Endcall Trnsfer Hold

max-calls-per-button 3

busy-trigger-per-button 2

!

!

ephone 12

ephone-template 5

mac-address 000F.9054.31BD

type 7960

button 1:10 2:7

What to Do Next

SCCP: Configuring Live Record

To configure the Live Record feature so that a phone user can record a conversation by pressing the LiveRcd soft key, perform the followings steps.

Prerequisites

Restrictions

  • Only one live record session is allowed for each conference.
  • Only the conference creator can initiate a live record session. In an ad hoc conference, participants who are not the conference creator cannot start a live record session. In a two-party call, the party who starts the live record session is the conference creator.

Note For legal disclaimer information about this feature, see DISCLAIMER: The use of monitoring, recording, or listening devices to eavesdrop, monitor, retrieve, or record phone conversations or other sound activities, whether or not contemporaneous with transmission, may be illegal in certain circumstances under federal, state and/or local laws. Legal advice should be sought prior to implementing any practice that monitors or records any phone conversation. Some laws require some form of notification to all parties to a phone conversation, such as by using a beep tone or other notification method or requiring the consent of all parties to the phone conversation, prior to monitoring or recording the phone conversation. Some of these laws incorporate strict penalties. In cases where local laws require a periodic beep while a conversation is being recorded, the Cisco Unity Express voice-mail system provides a user with the option of activating “the beep.” Prior to activating the Cisco Unity Express live record function, check the laws of all applicable jurisdictions. This is not legal advice and should not take the place of obtaining legal advice from a lawyer. IN ADDITION TO THE GENERAL DISCLAIMER THAT ACCOMPANIES THIS CISCO UNITY EXPRESS PRODUCT, CISCO ADDITIONALLY DISCLAIMS ANY AND ALL LIABILITY, BOTH CIVIL AND CRIMINAL, AND ASSUMES NO RESPONSIBILITY FOR THE UNAUTHORIZED AND/OR ILLEGAL USE OF THIS CISCO UNITY EXPRESS PRODUCT. THIS DISCLAIMER OF LIABILITY INCLUDES, BUT IS NOT NECESSARILY LIMITED TO, THE UNAUTHORIZED AND/OR ILLEGAL RECORDING AND MONITORING OF TELEPHONE CONVERSATIONS IN VIOLATION OF APPLICABLE FEDERAL, STATE AND/OR LOCAL LAWS..


SUMMARY STEPS

1. enable

2. configure terminal

3. telephony-service

4. live-record number

5. voicemail number

6. exit

7. ephone-dn dn-tag

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

9. call-forward all target-number

10. exit

11. ephone-template template-tag

12. softkeys connected {[ Acct ] [ ConfList ] [ Confrn ] [ Endcall ] [ Flash ] [ HLog ] [ Hold ] [ Join ] [ LiveRcd ] [ Park ] [ RmLstC ] [ Select ] [ TrnsfVM ] [ Trnsfer ]}

13. exit

14. ephone phone-tag

15. ephone-template template-tag

16. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

telephony-service

 

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 4

live record number

 

Router(config-telephony)# live record 8900

Defines the extension number that is dialed when the LiveRcd soft key is pressed on an SCCP IP phone.

Step 5

voicemail number

 

Router(config-telephony)# voicemail 8000

Defines the extension number that is speed-dialed when the Messages button is pressed on an IP phone.

  • Number —Cisco Unity Express voice-mail pilot number.

Step 6

exit

 

Router(config-telephony)# exit

Exits telephony-service configuration mode.

Step 7

ephone-dn dn-tag

 

Router(config)# ephone-dn 10

Creates a directory number that forwards all calls to the Cisco Unity Express voice-mail pilot number.

Step 8

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

 

Router(config-ephone-dn)# number 8900

Assigns an extension number to this directory number.

  • Number —Must match the Live Record pilot-number configured in Step 4.

Step 9

call-forward all target-number

 

Router(config-ephone-dn)# call-forward all 8000

Forwards all calls to this extension to the specified voice-mail number.

  • target-number —Phone number to which calls are forwarded. Must match the voice-mail pilot number configured in Step 5.

Note Phone users can activate and cancel the call-forward-all state from the phone using the CFwdAll soft key or a FAC.

Step 10

exit

 

Router(config-ephone-dn)# exit

Exits ephone-dn configuration mode.

Step 11

ephone-template template-tag

 

Router(config)# ephone-template 5

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

  • template-tag —Unique identifier for the ephone template. Range: 1 to 20.

Step 12

softkeys connected {[ Acct ] [ ConfList ] [ Confrn ] [ Endcall ] [ Flash ] [ HLog ] [ Hold ] [ Join ] [ LiveRcd ] [ Park ] [ RmLstC ] [ Select ] [ TrnsfVM ] [ Trnsfer ]}

 

Router(config-ephone-template)# softkeys connected LiveRcd Confrn Hold Park Trnsfer TrnsfVM

Modifies the order and type of soft keys that display on an IP phone during the connected call state.

Step 13

exit

 

Router(config-ephone-template)# exit

Exits ephone-template configuration mode.

Step 14

ephone phone-tag

 

Router(config)# ephone 12

Enters ephone configuration mode.

  • phone-tag —Unique number that identifies this ephone during configuration tasks.

Step 15

ephone-template template-tag

 

Router(config-ephone)# ephone-template 5

Applies the ephone template to the phone.

  • template-tag —Unique identifier of the ephone template that you created in Step 11.

Step 16

end

 

Router(config-ephone)# end

Exits to privileged EXEC mode.

Example

The following example shows Live Record is enabled at the system-level for extension 8900. All incoming calls to extension 8900 are forwarded to the voice-mail pilot number 8000 when the LiveRcd soft key is pressed, as configured under ephone-dn 10. Ephone template 5 modifies the display order of the LiveRcd soft key on IP phones.

telephony-service

privacy-on-hold

max-ephones 100

max-dn 240

timeouts transfer-recall 60

live-record 8900

voicemail 8000

max-conferences 8 gain -6

transfer-system full-consult

fac standard

!

!

ephone-template 5

softkeys remote-in-use CBarge Newcall

softkeys hold Resume Newcall Join

softkeys connected LiveRcd Confrn Hold Park Trnsfer TrnsfVM

max-calls-per-button 3

busy-trigger-per-button 2

!

!

ephone-dn 10

number 8900

call-forward all 8000

SIP: Configuring a Voice Mailbox Pilot Number

To configure the telephone number that is speed-dialed when the Message button on a SIP phone is pressed, follow the steps in this section.


NoteThe same telephone number is configured for voice messaging for all SIP phones in Cisco Unified CME. The The same telephone number is configured for voice messaging for all SIP phones in Cisco Unified CME. The call forward b2bua command enables call forwarding and designates that calls that are forwarded to a busy or no-answer extension be sent to a voicemail box.


Prerequisites

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register global

4. voicemail phone-number

5. exit

6. voice register dn dn-tag

7. call-forward b2bua busy directory-number

8. call-forward b2bua mailbox directory-number

9. call-forward b2bua noan directory-number

10. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register global

 

Router(config)# voice register global

Enters voice register global configuration mode to set parameters for all supported SIP phones in Cisco Unified CME.

Step 4

voicemail phone-number

 

Router(config-register-global)# voice mail 1111

Defines the telephone number that is speed-dialed when the Messages button on a Cisco Unified IP phone is pressed.

  • phone-number— Same phone number is configured for voice messaging for all SIP phones in a Cisco Unified CME.

Step 5

exit

 

Router(config-register-global)# exit

Exits voice register global configuration mode.

Step 6

voice register dn dn-tag

 

Router(config)# voice register dn 2

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

Step 7

call-forward b2bua busy directory-number

 

Router(config-register-dn)# call-forward b2bua busy 1000

Enables call forwarding for a SIP back-to-back user agent so that incoming calls to an extension that is busy will be forwarded to the designated directory number.

Step 8

call-forward b2bua mailbox directory-number

 

Router(config-register-dn)# call-forward b2bua mailbox 2200

Designates the voice mailbox to use at the end of a chain of call forwards.

  • Incoming calls have been forwarded to a busy or no-answer extension will be forwarded to the directory-number specified.

Step 9

call-forward b2bua noan directory-number timeout seconds

 

Router(config-register-dn)# call-forward b2bua noan 2201 timeout 15

Enables call forwarding for a SIP back-to-back user agent so that incoming calls to an extension that does not answer will be forwarded to the designated directory number.

  • seconds —Number of seconds that a call can ring with no answer before the call is forwarded to another extension. Range: 3 to 60000. Default: 20.

Step 10

end

 

Router(config-register-dn)# end

Exits to privileged EXEC mode.

What to Do Next

Enabling DTMF Integration

Perform one of the following tasks, depending on which DTMF-relay method is required:

Enabling DTMF Integration for Analog Voice-Mail Applications

To set up DTMF integration patterns for analog voice-mail applications, perform the following steps.


NoteYou can configure multiple tags and tokens for each pattern, depending on the voice-mail system and type of access. You can configure multiple tags and tokens for each pattern, depending on the voice-mail system and type of access.


SUMMARY STEPS

1. enable

2. configure terminal

3. vm-integration

4. pattern direct tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

5. pattern ext-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [last-tag ]

6. pattern ext-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

7. pattern trunk-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

8. pattern trunk-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }]
[
tag3 { CGN | CDN | FDN }] [ last-tag ]

9. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

vm-integration

 
Router(config) vm-integration

Enters voice-mail integration configuration mode and enables voice-mail integration with DTMF and an analog voice-mail system.

Step 4

pattern direct tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-integration) pattern direct 2 CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone.

  • The tag attribute is an alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A, B, C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system’s integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
  • The keywords, CGN , CDN , and FDN , configure the type of call information sent to the voice-mail system, such as calling number (CGN), called number (CDN), or forwarding number (FDN).

Step 5

pattern ext-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-integration) pattern ext-to-ext busy 7 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail.

Step 6

pattern ext-to-ext no-answer tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-integration) pattern ext-to-ext no-answer 5 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an internal extension fails to connect to an extension and the call is forwarded to voice mail.

Step 7

pattern trunk-to-ext busy tag1 { CGN | CDN | FDN } [ tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-integration) pattern trunk-to-ext busy 6 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches a busy extension and the call is forwarded to voice mail.

Step 8

pattern trunk-to-ext no-answer tag1 { CGN | CDN | FDN } [tag2 { CGN | CDN | FDN }] [ tag3 { CGN | CDN | FDN }] [ last-tag ]

 

Router(config-vm-integration)# pattern trunk-to-ext no-answer 4 FDN * CGN *

Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.

Step 9

end

 

Router(config-vm-integration)# exit

Exits configuration mode and enters privileged EXEC mode.

What to Do Next

After configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI) notification for either the MWI outcall, unsolicited notify, or subscribe/notify mechanism. See the “SCCP: Configuring a Phone for MWI Outcall” section.

Enabling DTMF Integration Using RFC 2833

To configure a SIP dial peer to point to Cisco Unity and enable SIP dual-tone multifrequency (DTMF) relay using RFC 2833, use the commands in this section on both the originating and terminating gateways.

This DTMF relay method is required in the following situations:

  • When SIP is used to connect Cisco Unified CME to a remote SIP-based IVR or voice-mail application such as Cisco Unity.
  • When SIP is used to connect Cisco Unified CME to a remote SIP-PSTN voice gateway that goes through the PSTN to a voice-mail or IVR application.

NoteIf the T.38 Fax Relay feature is also configured on this IP network, we recommend that you either configure the voice gateways to use a payload type other than PT96 or PT97 for fax relay negotiation, or depending on whether the SIP endpoints support different payload types, configure Cisco Unified CME to use a payload type other than PT96 or PT97 for DTMF. If the T.38 Fax Relay feature is also configured on this IP network, we recommend that you either configure the voice gateways to use a payload type other than PT96 or PT97 for fax relay negotiation, or depending on whether the SIP endpoints support different payload types, configure Cisco Unified CME to use a payload type other than PT96 or PT97 for DTMF.


Prerequisites

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. description string

5. destination-pattern string

6. session protocol sipv2

7. session target { dns : address | ipv4 : destination-address }

8. dtmf-relay rtp-nte

9. dtmf-interworking rtp-nte

10. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

dial-peer voice tag voip

 

Router (config)# dial-peer voice 123 voip

Enters dial-peer configuration mode to define a VoIP dial peer for the voice-mail system.

  • tag —Defines the dial peer being configured. Range is 1 to 2147483647.

Step 4

description string

 

Router (config-voice-dial-peer)# description CU pilot

(Optional) Associates a description with the dial peer being configured. Enter a string of up to 64 characters.

Step 5

destination-pattern string

 

Router (config-voice-dial-peer)# destination-pattern 20

Specifies the pattern of the numbers that the user must dial to place a call.

  • string —Prefix or full E.164 number.

Step 6

session protocol sipv2

 

Router (config-voice-dial-peer)# session protocol sipv2

Specifies that Internet Engineering Task Force (IETF) Session Initiation Protocol (SIP) is protocol to be used for calls between local and remote routers using the packet network.

Step 7

session target { dns : address | ipv4 : destination-address }

 

Router (config-voice-dial-peer)# session target ipv4:10.8.17.42

Designates a network-specific address to receive calls from the dial peer being configured.

  • dns : address —Specifies the DNS address of the voice-mail system.
  • ipv4 : destination- address —Specifies the IP address of the voice-mail system.

Step 8

dtmf-relay rtp-nte

 

Router (config-voice-dial-peer)# dtmf-relay rtp-nte

Sets DTMF relay method for the voice dial peer being configured.

  • rtp-nte — Provides conversion from the out-of-band SCCP indication to the SIP standard for DTMF relay (RFC 2833). Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with the Named Telephone Event (NTE) payload type.
  • This command can also be configured in voice-register-pool configuration mode. For individual phones, the phone-level configuration for this command overrides the system-level configuration for this command.

Note The need to use out-of-band conversion is limited to SCCP phones. SIP phones natively support in-band.

Step 9

dtmf-interworking rtp-nte

 

Router (config-voice-dial-peer)# dtmf-interworking rtp-nte

(Optional) Enables a delay between the dtmf-digit begin and dtmf-digit end events in the RFC 2833 packets.

  • This command is supported in Cisco IOS Release 12.4(15)XZ and later releases and in Cisco Unified CME 4.3 and later versions.
  • This command can also be configured in voice-service configuration mode.

Step 10

end

 

Router(config-voice-dial-peer)# end

Exits to privileged EXEC mode.

What to Do Next

After configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI) notification for either the MWI outcall, unsolicited notify, or subscribe/notify mechanism. See the “SCCP: Configuring a Phone for MWI Outcall” section.

Enabling DTMF Integration Using SIP NOTIFY

To configure a SIP dial peer to point to Cisco Unity Express and enable SIP dual-tone multifrequency (DTMF) relay using SIP NOTIFY format, follow the steps in this task.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. description string

5. destination-pattern string

6. b2bua

7. session protocol sipv2

8. session target { dns : address | ipv4 : destination-address }

9. dtmf-relay sip-notify

10. codec g711ulaw

11. no vad

12. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal#

Enters global configuration mode.

Step 3

dial-peer voice tag voip

 

Router (config)# dial-peer voice 2 voip

Enters dial-peer configuration mode to define a VoIP dial peer for the voice-mail system.

  • tag —Defines the dial peer being configured. Range is 1 to 2147483647.

Step 4

description string

 

Router (config-voice-dial-peer)# description cue pilot

(Optional) Associates a description with the dial peer being configured. Enter a string of up to 64 characters.

Step 5

destination-pattern string

 

Router (config-voice-dial-peer)# destination-pattern 20

Specifies the pattern of the numbers that the user must dial to place a call.

  • string —Prefix or full E.164 number.

Step 6

b2bua

 

Router (config-voice-dial-peer)# b2bua

(Optional) Includes the Cisco Unified CME address as part of contact in 3XX response to point to Cisco Unity Express and enables SIP-to-SCCP call forward.

Step 7

session protocol sipv2

 

Router (config-voice-dial-peer)# session protocol sipv2

Specifies that Internet Engineering Task Force (IETF) Session Initiation Protocol (SIP) is protocol to be used for calls between local and remote routers using the packet network.

Step 8

session target { dns : address | ipv4 : destination-address }

 

Router (config-voice-dial-peer)# session target ipv4:10.5.49.80

Designates a network-specific address to receive calls from the dial peer being configured.

  • dns : address —Specifies the DNS address of the voice-mail system.
  • ipv4 : destination- address —Specifies the IP address of the voice-mail system.

Step 9

dtmf-relay sip-notify

 

Router (config-voice-dial-peer)# dtmf-relay sip-notify

Sets the DTMF relay method for the voice dial peer being configured.

  • sip-notify — Forwards DTMF tones using SIP NOTIFY messages.
  • This command can also be configured in voice-register-pool configuration mode. For individual phones, the phone-level configuration for this command overrides the system-level configuration for this command.

Step 10

codec g711ulaw

 

Router (config-voice-dial-peer)# codec g711ulaw

Specifies the voice coder rate of speech for a dial peer being configured.

Step 11

no vad

 

Router (config-voice-dial-peer)# no vad

Disables voice activity detection (VAD) for the calls using the dial peer being configured.

Step 12

end

 

Router(config-voice-dial-peer)# end

Exits to privileged EXEC mode.

What to Do Next

After configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI). See the “SCCP: Configuring a Phone for MWI Outcall” section.

SCCP: Configuring a Phone for MWI Outcall

To designate a phone line or directory number on an individual SCCP phone to be monitored for voice-mail messages, or to enable audible MWI, perform the following steps.

Prerequisites

Restrictions

  • Audible MWI is supported only in Cisco Unified CME 4.0(2) and later versions.
  • Audible MWI is supported only on Cisco Unified IP Phone 7931G and Cisco Unified IP Phone 7911.

SUMMARY STEPS

1. enable

2. configure terminal

3. ephone phone-tag

4. mwi-line line-number

5. exit

6. ephone-dn dn-tag

7. mwi { off | on | on-off }

8. mwi-type { visual | audio | both }

9. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

ephone phone-tag

 

Router(config)# ephone 36

Enters ephone configuration mode.

Step 4

mwi-line line-number

 

Router(config-ephone)# mwi-line 3

(Optional) Selects a phone line to receive MWI treatment.

  • line-number —Number of phone line to receive MWI notification. Range: 1 to 34. Default: 1.

Step 5

exit

 

Router(config-ephone)# exit

Exits ephone configuration mode.

Step 6

ephone-dn dn-tag

 

Router(config)# ephone-dn 11

Enters ephone-dn configuration mode.

Step 7

mwi { off | on | on-off }

 

Router(config-ephone-dn)# mwi on-off

(Optional) Enables a specific directory number to receive MWI notification from an external voice-messaging system.

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

Step 8

mwi-type { visual | audio | both }

 

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

(Optional) Specifies which type of MWI notification to be received.

Note This command is supported only on the Cisco Unified IP Phone 7931G and Cisco Unified IP Phone 7911.

Note This command can also be configured in ephone-dn-template configuration mode. The value that you set in ephone-dn configuration mode has priority over the value set in ephone-dn-template mode. For configuration information, see SCCP: Enabling Ephone-dn Templates.

Step 9

end

 

Router(config-ephone-dn)# end

Returns to privileged EXEC mode.

SIP: Enabling MWI at the System-Level

To enable a message waiting indicator (MWI) at a system-level, perform the following steps.

Prerequisites

  • Cisco CME 3.4 or a later version.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register global

4. mwi reg-e164

5. mwi stutter

6. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register global

 

Router(config)# voice register global

Enters voice register global configuration mode to set parameters for all supported SIP phones in Cisco Unified CME.

Step 4

mwi reg-e164

 

Router(config-register-global)# mwi reg-e164

Registers full E.164 number to the MWI server in Cisco Unified CME and enables MWI.

Step 5

mwi stutter

 

Router(config-register-global)# mwi stutter

Enables Cisco Unified CME router at the central site to relay MWI notification to remote SIP phones.

Step 6

end

 

Router(config-register-global)# end

Exits to privileged EXEC mode.

SIP: Configuring a Directory Number for MWI

Perform one of the following tasks, depending on whether you want to configure MWI outcall or MWI notify (unsolicited notify or subscribe/notify) for SIP endpoints in Cisco Unified CME.

SIP: Defining Pilot Call Back Number for MWI Outcall

To designate a phone line on an individual SIP directory number to be monitored for voice-mail messages, perform the following steps.

Prerequisites

Restrictions

  • For Cisco Unified CME 4.1 and later versions, the Call Forward All, Presence, and MWI features require that SIP phones must be configured with a directory number by using the number command with the dn keyword; direct line numbers are not supported.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register dn dn-tag

4. mwi

5. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register dn dn-tag

 

Router(config)# 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

mwi

 

Router(config-register-dn)# mwi

Enables a specific directory number to receive MWI notification.

Step 5

end

 

Router(config-ephone-dn)# end

Exits to privileged EXEC mode.

SIP: Configuring a Directory Number for MWI NOTIFY

To identify the MWI server and specify a directory number for receiving MWI Subscribe/NOTIFY or MWI Unsolicited NOTIFY, follow the steps in this section.


NoteWe recommend using the Subscribe/NOTIFY method instead of an Unsolicited NOTIFY when possible. We recommend using the Subscribe/NOTIFY method instead of an Unsolicited NOTIFY when possible.


Prerequisites

Restrictions

  • For Cisco Unified CME 4.1 and later versions, the Call Forward All, Presence, and MWI features require that SIP phones must be configured with a directory number by using the number command with the dn keyword; direct line numbers are not supported.
  • The SIP MWI - QSIG Translation feature in Cisco Unified CME 4.1 does not support Subscribe NOTIFY.
  • Cisco Unified IP Phone 7960, 7940, 7905, and 7911 support only Unsolicited NOTIFY for MWI.

SUMMARY STEPS

1. enable

2. configure terminal

3. sip-ua

4. mwi-server { ipv4 : destination-address | dns : host-name } [ unsolicited ]

5. exit

6. voice register dn dn-tag

7. mwi

8. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

sip-ua

 

Router(config)# sip-ua

Enters Session Initiation Protocol (SIP) user agent (ua) configuration mode for configuring the user agent.

Step 4

mwi-server { ipv4 : destination-address | dns : host-name } [ unsolicited ]

 

Router(config-sip-ua)# mwi-server ipv4:1.5.49.200

or

Router(config-sip-ua)# mwi-server dns:server.yourcompany.com unsolicited

Specifies voice-mail server settings on a voice gateway or UA.

Note The sip-server and mwi expires commands under the telephony-service configuration mode have been migrated to mwi-server to support DNS format of the SIP server.

Step 5

exit

 

Router(config-sip-ua)# exit

Exits to the next highest mode in the configuration mode hierarchy.

Step 6

voice register dn dn-tag

 

Router(config)# 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 7

mwi

 

Router(config-register-dn)# mwi

Enables a specific directory number to receive MWI notification.

Step 8

end

 

Router(config-register-dn)# end

Exits to privileged EXEC mode.

Enabling SIP MWI Prefix Specification

To accept unsolicited SIP Notify messages for MWI that include a prefix string as a site identifier, perform the following steps.

Prerequisites

SUMMARY STEPS

1. enable

2. telephony-service

3. mwi prefix prefix-string

4. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

telephony-service

 

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 3

mwi prefix prefix-string

 

Router(config-telephony)# mwi prefix 555

Specifies a string of digits that, if present before a known Cisco Unified CME extension number, are recognized as a prefix.

  • prefix-string —Digit string. The maximum prefix length is 32 digits.

Step 4

end

 

Router(config-telephony)# end

Returns to privileged EXEC mode.

SIP: Configuring VMWI

To enable a VMWI, perform the following steps.

Prerequisites

  • Cisco IOS Release 12.4(6)T or a later version

SUMMARY STEPS

1. enable

2. configure terminal

3. voice-port port

4. mwi

5. vmwi dc-voltage

or

vmwi fsk

6. exit

7. sip-ua

8. mwi-server { ipv4 : destination-address | dns : host-name } [ unsolicited ]

9. end

DETAILED STEPS

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice-port port

 

Router(config)# voice-port 2/0

Enters voice-port configuration mode.

  • port —Syntax is platform-dependent. Type ? to determine.

Step 4

mwi

 

Router(config-voiceport)# mwi

Enables MWI for a specified voice port.

Step 5

vmwi dc-voltage

 

or

vmwi fsk

 

Router(config-voiceport)# vmwi dc-voltage

(Optional) Enables DC voltage or FSK VMWI on a Cisco VG224 onboard analog FXS voice port.

You do not need to perform this step for the Cisco VG202 and Cisco VG204. They support FSK only. VMWI is configured automatically when MWI is configured on the voice port.

This step is required for the VG224. If an FSK phone is connected to the voice port, use the fsk keyword. If a DC voltage phone is connected to the voice port, use the dc-voltage keyword.

Step 6

exit

 

Router(config-sip-ua)# exit

Exits to the next highest mode in the configuration mode hierarchy.

Step 7

sip-ua

 

Router(config)# sip-ua

Enters Session Initiation Protocol user agent configuration mode for configuring the user agent.

Step 8

mwi-server { ipv4 : destination-address | dns : host-name } [ unsolicited ]

 

Router(config-sip-ua)# mwi-server ipv4:1.5.49.200

or

Router(config-sip-ua)# mwi-server dns:server.yourcompany.com unsolicited

Specifies voice-mail server settings on a voice gateway or user agent (ua).

Note The sip-server and mwi expires commands under the telephony-service configuration mode have been migrated to mwi-server to support DNS format of the Session Initiation Protocol (SIP) server.

Step 9

end

 

Router(config-voiceport)# end

Exits voice-port configuration mode and returns to privileged EXEC mode.

 

Verifying Voice-Mail Integration

  • Press the Messages button on a local phone in Cisco Unified CME and listen for the voice mail greeting.
  • Dial an unattended local phone and listen for the voice mail greeting.
  • Leave a test message.
  • Go to the phone that you called. Verify that the [Message] indicator is lit.
  • Press the Messages button on this phone and retrieve the voice mail message.

Configuration Examples for Voice-Mail Integration

This section contains the following examples:

Mailbox Selection Policy for SCCP Phones: Example

The following example sets a policy to select the mailbox of the originally called number when a call is diverted to a Cisco Unity Express or PBX voice-mail system with the pilot number 7000.

dial-peer voice 7000 voip

destination-pattern 7000

session target ipv4:10.3.34.211

codec g711ulaw

no vad

mailbox-selection orig-called-num

The following example sets a policy to select the mailbox of the last number that the call was diverted to before being diverted to a Cisco Unity voice-mail system with the pilot number 8000.

ephone-dn 825

number 8000

mailbox-selection last-redirect-num

Voice Mailbox for SIP Phones: Example

The following example shows how to configure the call forward b2bua mailbox for SIP endpoints:

voice register global

voicemail 1234

!

voice register dn 2

number 2200

call-forward b2bua all 1000

call-forward b2bua mailbox 2200

call-forward b2bua noan 2201 timeout 15

mwi

DTMF Integration Using RFC 2833: Example

The following example shows the configuration for DTMF Relay using RFC 2833:

dial-peer voice 1 voip

destination-pattern 4…

session target ipv4:10.8.17.42

session protocol sipv2

dtmf-relay sip-notify rtp-nte

DTMF Integration Using SIP Notify: Example

The following example shows the configuration for DTMF using SIP Notify:

dial-peer voice 1 voip

destination-pattern 4…

session target ipv4:10.5.49.80

session protocol sipv2

dtmf-relay sip-notify

b2bua

DTMF Integration for Legacy Voice-Mail Applications: Example

The following example sets up DTMF integration for an analog voice-mail system.

vm-integration

pattern direct 2 CGN *

pattern ext-to-ext busy 7 FDN * CGN *

pattern ext-to-ext no-answer 5 FDN * CGN *

pattern trunk-to-ext busy 6 FDN * CGN *

pattern trunk-to-ext no-answer 4 FDN * CGN *

SCCP Phone Line for MWI: Example

The following example enables MWI on ephone 18 for line 2 (button 2), which has overlaid ephone-dns. Only a message waiting for the first ephone-dn (2021) on this line will activate the MWI lamp. Button 4 is unused. The line numbers in this example are as follows:

  • Line 1—Button 1—Extension 2020
  • Line 2—Button 2—Extension 2021, 2022, 2023, 2024
  • Line 3—Button 3—Extension 2021, 2022, 2023, 2024 (rollover line)
  • Button 4—Unused
  • Line 4—Button 5—Extension 2025

ephone-dn 20

number 2020

ephone-dn 21

number 2021

ephone-dn 22

number 2022

ephone-dn 23

number 2023

ephone-dn 24

number 2024

ephone-dn 25

number 2025

ephone 18

button 1:20 2o21,22,23,24,25 3x2 5:26

mwi-line 2

The following example enables MWI on ephone 17 for line 3 (extension 609). In this example, the button numbers do not match the line numbers because buttons 2 and 4 are not used. The line numbers in this example are as follows:

  • Line 1—Button 1—Extension 607
  • Button 2—Unused
  • Line 2—Button 3—Extension 608
  • Button 4—Unused
  • Line 3—Button 5—Extension 609

ephone-dn 17

number 607

ephone-dn 18

number 608

ephone-dn 19

number 609

ephone 25

button 1:17 3:18 5:19

mwi-line 3

SIP MWI Prefix Specification: Example

The following example identifies the SIP server for MWI notification at the IP address 172.16.14.22. It states that the Cisco Unified CME system will accept unsolicited SIP Notify messages for known mailbox numbers using the prefix 555.

sip-ua

mwi-server 172.16.14.22 unsolicited

telephony-service

mwi prefix 555

SIP Directory Number for MWI Outcall: Example

The following example shows an MWI callback pilot number:

voice register dn

number 9000….

mwi

SIP Directory Number for MWI Unsolicited Notify: Example

The following example shows how to specify voice-mail server settings on a UA. The example includes the unsolicited keyword, enabling the voice-mail server to send a SIP notification message to the UA if the mailbox status changes and specifies that voice dn 1, number 1234 on the SIP phone in Cisco Unified CME will receive the MWI notification:

sip-ua

mwi-server dns:server.yourcompany.com expires 60 port 5060 transport udp unsolicited

voice register dn 1

number 1234

mwi

SIP Directory Number for MWI Subscribe/NOTIFY: Example

The following example shows how to define an MWI server and specify that directory number 1, number 1234 on a SIP phone in Cisco Unified CME is to receive the MWI notification:

sip-ua

mwi-server ipv4:1.5.49.200

voice register dn 1

number 1234

mwi

Additional References

The following sections provide references related to Cisco Unified CME features.

Technical Assistance

Description
Link

The Cisco Support website provides extensive online resources, including documentation and tools for troubleshooting and resolving technical issues with Cisco products and technologies.

To receive security and technical information about your products, you can subscribe to various services, such as the Product Alert Tool (accessed from Field Notices), the Cisco Technical Services Newsletter, and Really Simple Syndication (RSS) Feeds.

Access to most tools on the Cisco Support website requires a Cisco.com user ID and password.

http://www.cisco.com/techsupport

Feature Information for Voice-Mail Integration

Table 1 lists the features in this module and enhancements to the features by version.

To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/33matrix.htm .

Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn . An account on Cisco.com is not required.


Note Table 1 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.


 

Table 1 Feature Information for Voice-Mail Integration

Feature Name
Cisco Unified CME
Version
Feature Information

Audible MWI

4.0(2)

Provides support for selecting audible, visual, or audible and visual Message Waiting Indicator (MWI) on supported Cisco Unified IP phones.

Cisco Unity Express AXL Enhancement

7.0(1)

Cisco Unified CME and Cisco Unity Express passwords are automatically synchronized. No configuration is required for this feature.

DTMF Integration

3.4

Added support for voice messaging systems connected via a SIP trunk or SIP user agent.

The standard Subscribe/NOTIFY method is preferred over an Unsolicited NOTIFY.

2.0

DTMF integration patterns were introduced.

Live Record

4.3

Enables IP phone users in a Cisco Unified CME system to record a phone conversation if Cisco Unity Express is the voice mail system.

Mailbox Selection Policy

4.0

Mailbox selection policy was introduced.

MWI

4.0

MWI line selection of a phone line other than the primary line on a SCCP phone was introduced.

3.4

Voice messaging systems (including Cisco Unity) connected via a SIP trunk or SIP user agent can pass a Message Waiting Indicator (MWI) that will be received and understood by a SIP phone directly connected to Cisco Unified CME.

SIP MWI Prefix Specification

4.0

SIP MWI prefix specification was introduced.

SIP MWI - QSIG Translation

4.1

Extends message waiting indicator (MWI) functionality for SIP MWI and QSIG MWI interoperation to enable sending and receiving of MWI over QSIG to PBX.

Transfer to Voice Mail

4.3

Enables a phone user to transfer a caller directly to a voice-mail extension.