Step 6: Configuring Cisco UE AA and Voice Mail

Step 6 Configuring Cisco UE AA and Voice Mail

If your system does not have Cisco UE installed, you can skip this section and proceed to Step 7. The following sections step you through setting up the basic Cisco UE configuration necessary to use the AA and voice mail on your system.

Setting Up the Router for Cisco UE

Before you can configure and use Cisco UE applications such as AA and voice mail, you must set up the following basic IP connectivity parameters to ensure that the Cisco UE software can communicate with its environment:

• IP addressing
• Call routing to Cisco UE
• H.323-to-Session Initiation Protocol (SIP) call routing
• Message Waiting Indicator (MWI)

IP Addressing

The Cisco UE hardware module must be configured with an IP address on the router. This setup is covered in Chapter 13. Following the information provided there, the resulting configuration for the Cisco UE module for Site A is shown in Example 15-18.

Example 15-18. Service-Engine Interface for Cisco UE

cme-3725#show running-config interface Service-Engine1/0 ip unnumbered FastEthernet0/0 service-module ip address 10.1.235.128 255.255.0.0 service-module ip default-gateway 10.1.235.1 ip route 10.1.235.128 255.255.255.255 Service-Engine1/0

You can show the status of the Cisco UE module by using the command shown in Example 15-19. You can also ping the IP interface (10.1.229.128 in this example) to ensure that IP connectivity is established.

Example 15-19. Showing the Status of the Cisco UE Module

cme-3725#service-module service-engine 1/0 status Service Module is Cisco Service-Engine1/0 Service Module supports session via TTY line 33 Service Module is in Steady state Getting status from the Service Module, please wait.. cisco service engine 1.1

Call Routing to Cisco UE

Cisco CME and Cisco UE communicate using a SIP interface. Therefore, Cisco CME uses SIP dial peers to determine which calls must be routed to Cisco UE. The AA pilot number for Site A is 2100, and the voice mail pilot is 2105. Example 15-20 shows the SIP dial peers necessary to route calls to these pilot numbers from Cisco CME to Cisco UE.

Example 15-20. SIP Dial Peers Needed for Call Routing

cme-3725#show running-config dial-peer voice 2100 voip destination-pattern 21.. session protocol sipv2 session target ipv4:10.1.235.128 dtmf-relay sip-notify codec g711ulaw no vad

H.323-to-SIP Call Routing

If you have multiple sites to connect to each other, such as Site B in the sample topology used in this chapter, calls between sites use H.323, and calls to Cisco UE use SIP. This requires an H.323-to-SIP translation on the Cisco CME router. The following CLI enables this function (available with Cisco CME 3.2 and later):

voice service voip allow-connections h323 to sip

Message Waiting Indicator

Chapter 10, "Cisco IPC Express Integrated Voice Mail," explains the mechanism for turning MWI on and off from Cisco UE voice mail to Cisco CME phones. The MWI definitions shown in Example 15-21 are necessary to enable MWI for Site A.

Example 15-21. MWI DN Definition

cme-3725#show running-config ephone-dn 51 number 8000.... mwi on ! ephone-dn 52 number 8001.... mwi off

You are now ready to point a browser to Cisco UE (http://10.1.235.128/ for Site A) to run through the Cisco UE Initialization Wizard. This is covered in the next section.

Configuring Basic Cisco UE

The Cisco UE Initialization Wizard, covered in Chapter 13, allows you to import the configuration (primarily phones and extensions) already done on your Cisco CME system into Cisco UE, create mailboxes for the users, and define the pilot numbers for the AA and voice mail.

Importing Users from Cisco CME

Figure 15-4 shows how the six Site B users already defined on the router are imported into Cisco UE in the first Initialization Wizard screen.

Figure 15-4. Importing Users

Check the boxes in the Mailbox column for all six users to automatically create personal mailboxes for all six users at the end of the Initialization Wizard.

Setting System Defaults

In the System Defaults screen, you set attributes such as the PIN and password generation policy for new accounts. Figure 15-5 shows the settings and selections made for the Site B configuration.

Figure 15-5. Setting System Defaults

Setting Call Handling Parameters

The pilot numbers for the AA (3101), voice mail (3105), and the administration TUI (3106) are set in the Call Handling screen, shown in Figure 15-6. Also, the MWI DNs imported from the Cisco CME definitions entered earlier (refer to Example 15-21) are imported into this screen.

Figure 15-6. Setting Call Handling Parameters

At this point in the system setup, you can call into the system AA and voice mail.

Configuring Voice Mail

Figure 15-7 shows the final screen of the Initialization Wizard. It summarizes the auto-generated system passwords and PINs for all the user profiles and mailboxes created by the Initialization Wizard. This is a handy screen to preserve as a screenshot or printout to help you let each user know his default password and PIN for first-time login to his mailbox.

Figure 15-7. Cisco UE User Passwords and PINs

All users have mailboxes with the tutorial set to yes. When you notify users of their default system-assigned PINs, they can log into their mailboxes and work through the setup tutorial. The tutorial helps them record a spoken name and an outgoing greeting and forces them to change their PIN to a private setting not known to you as the system administrator.

Calls to your employees' phones automatically forward into voice mail, where the caller hears the standard system greetings until your employees have logged in to customize their mailbox greetings. Call forwarding was set up in Step 3, earlier in this chapter, by the Cisco CME Setup Utility. Example 15-22 shows the call forwarding setup for ephone-dns. If you want to change this call forward destination, go to the Configure > Extensions GUI screen or use the ephone-dn command in the CLI.

Example 15-22. Call Forward Setup for Ephone-dns

cme-3725#show running-config ephone-dn 1 dual-line number 2001 secondary 2225552001 call-forward busy 2105 call-forward noan 2105 timeout 10

In the current configuration, PSTN calls cannot get into voice mail. Instead, callers hear, "Sorry, there is no mailbox associated with this extension," even though internal calls from other IP phones get into voice mail correctly. One way to address this is to add the direct inward dial (DID) number associated with the extension (and therefore the mailbox) to the Primary E.164 Number field in the Configure > Users screen, as shown in Figure 15-8. Now PSTN calls also work into voice mail. Alternatively, you can use Cisco IOS translation rules or Cisco CME dial plan patterns to translate the DID numbers to extensions before the calls enter voice mail.

Figure 15-8. Configuring the DID Number for a Mailbox

If you did not add mailboxes in the Initialization Wizard, go to the Configuration > Users GUI screen to add user definitions and mailboxes for the employees on your system.

At this point, basic voice mail is set up and working. Example 15-23 summarizes the mailboxes defined on the system and the time used in each.

Example 15-23. Mailbox Summary

[View full width]

cue-3725#show voicemail mailboxes OWNER MSGS NEW SAVED MSGTIME MBXSIZE USED "adavidson" 1 0 1 13 5520 1 % "awilkins" 1 0 1 9 5520 1 % "awyant" 0 0 0 5 5520 1 % "amcdougal" 0 0 0 5 5520 1 % "acoley" 0 0 0 5 5520 1 %

Configuring the AA

Cisco UE ships with a system AA that is very easy to set up. It offers callers dial-by-extension, dial-by-name, and transfer-to-the-operator choices. You can also install a fully customized AA into Cisco UE so that you can tailor the AA menus and choices to your own business needs.

Setting Up the System AA

If your office intends to use only the Cisco UE system AA, follow the steps in this section; otherwise, proceed to the next section.

The system AA is set up by default and is working already with the AA pilot number you assigned during the Cisco UE Initialization Wizard. Dial-by-extension from the system AA works without any further setup, but dial-by-name requires that you configure the names of your employees in their user profiles. In the Configure > Users GUI screen, click each user to see his or her profile, and fill in the First Name and Last Name fields with the strings you want to be used for matching in the AA dial-by-name feature.

The last remaining field to set up in the system AA is to customize your company's AA welcome greeting. Record this greeting offline on a PC, and upload the .wav file to Cisco UE. You also can record the greeting by using the administrative TUI (extension 2106 for Site A or 3106 for Site B) on Cisco UE. Associate the .wav file with the system AA.

The sample system setup in this chapter uses Cisco UE 2.0, and you have now configured all the system AA parameters for this release. If you are using Cisco UE release 2.1 (or later), a few more parameters need to be customized, as discussed in Chapter 14.

Setting Up a Custom AA

If your business requires a more sophisticated AA than the Cisco UE system AA, use the guidelines in this section to set up your custom AA. Chapter 9, "Cisco IPC Express Automated Attendant Options," explained how to customize an AA script on Cisco UE. Assume at Site B that you require only the system AA, whereas a custom AA is necessary at Site A.

For example, the script written for Site A is named S1_Main-OfficeHours.aef, and it calls two subflows: S1_DialbyExtension.aef and S1_XfertoOper.aef. You can download these scripts and the prompts they use from Cisco.com by going to the Software Center for Cisco UE. The S1_Main-OfficeHours script includes nine different prompts that you record on a PC or in a studio as a .wav file. Upload all three scripts (.aef files) and nine prompts (.wav files) from your PC to Cisco UE, as shown in Figure 15-9.

Figure 15-9. Uploading Scripts and Prompts to Cisco UE

As soon as the scripts and prompt files are available on the Cisco UE system, you have to add a custom AA to the system. Start this activity by going to the Voice mail > Auto Attendant GUI screen, and choose Add. Select the S1_Main-OfficeHours.aef script for the AA, as shown in Figure 15-10. Insert a name for the AA (the example uses custom-aa).

Figure 15-10. Selecting a Script for the Custom AA

Assign a pilot number for the AA, as shown in Figure 15-11. Because your real AA pilot number, extension 2100, is already assigned to the system AA, you cannot choose that. For the time being, choose 2101.

Figure 15-11. Choosing a Pilot Number for the Custom AA

After you add the AA, you see custom-aa show up in the Voice mail > Auto Attendant GUI screen. To switch around the pilot numbers, assuming that 2100 is the actual AA pilot number your business wants to use, click the system AA (autoattendant), and change its pilot number to a different extension (for example, 2102). Click custom-aa and change its pilot number to 2100. The resulting configuration is shown in Figure 15-12. Your custom AA is now operational.

Figure 15-12. Correcting the Pilot Number for the Custom AA

At this point, internal calls from the IP phones to the AA (extension 2100) work, but PSTN DID calls to the AA don't work. PSTN calls arrive at 222.555.2001, and this DID number must be mapped to the AA. Insert the Cisco IOS translation rule (or its equivalent), shown in Example 15-24, into the Cisco CME router, and attach it to the dial peer in Cisco UE that matches PSTN trunk calls. This same translation rule includes translations to allow calls into your voice mail (2105) and administrative TUI (2106) pilot numbers from the PSTN.

Example 15-24. Translation Rule to Route PSTN Calls to the AA

cme-3725#show running-config voice translation-rule 10 rule 1 /2225552100/ /2100/ rule 2 /2225552105/ /2105/ rule 3 /2225552106/ /2106/ ! voice translation-profile to_cue translate called 10 ! dial-peer voice 2100 voip destination-pattern 21.. session protocol sipv2 session target ipv4:10.1.235.128 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 2101 voip description VM-AA-PSTN translation-profile outgoing to_cue destination-pattern 22255521.. session protocol sipv2 session target ipv4:10.1.235.128 dtmf-relay sip-notify codec g711ulaw no vad

In the section "Routing PSTN Calls to IP Phones," the FXO PSTN trunk was set up to ring employee extension 2001, because the AA was not yet implemented. Now that the AA is fully configured, the routing of the FXO calls should be changed to the AA (extension 2100) instead of IP phone 2001:

voice-port 2/0/1 connection plar opx 2100