Unlock Iewb Vo Vol i v3.Section.1.Cucme Cue.1.6

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CCIE Voice Lab Workbook Volume I Version 3.0 Communications Manager Express

Copyright 2009 Internetwork Expert INE.comi

Copyright Information

Copyright 2009 Internetwork Expert, Inc. All rights reserved.

The following publication, CCIE Voice Lab Workbook Volume I Version 3.0, was developed by InternetworkExpert, Inc. All rights reserved. No part of this publication may be reproduced or distributed in any form or byany means without the prior written permission of Internetwork Expert, Inc.

Cisco, Cisco Systems, CCIE, and Cisco Certified Internetwork Expert, are registered trademarks ofCisco Systems, Inc. and/or its affiliates in the U.S. and certain countries.

All other products and company names are the trademarks, registered trademarks, and service marks of therespective owners. Throughout this manual, Internetwork Expert, Inc. has used its best efforts to distinguishproprietary trademarks from descriptive names by following the capitalization styles used by themanufacturer.


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Copyright 2009 Internetwork Expert INE.comii

Disclaimer

The following publication, CCIE Voice Lab Workbook Volume I Version 3.0, is designed to assist candidatesin the preparation for Cisco Systems CCIE Voice Lab Exam. While every effort has been made to ensurethat all material is as complete and accurate as possible, the enclosed material is presented on an as isbasis. Neither the authors nor Internetwork Expert, Inc. assume any liability or responsibility to any person

or entity with respect to loss or damages incurred from the information contained in this workbook.

This workbook was developed by Internetwork Expert, Inc. and is an original work of the aforementionedauthors. Any similarities between material presented in this workbook and actual CCIE lab material iscompletely coincidental.


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Copyright 2009 Internetwork Expert INE.comiii

Table of Contents

Cisco Unified Communications Manager Express ............................. 11.1 CUCME Basic Configuration.............................................................. 1

1.2 Phone Registration & Number Assignment (SCCP) .......................... 11.3 SIP Phones ........................................................................................ 21.4 ISDN PRI ........................................................................................... 21.5 IOS Call Routing ................................................................................ 21.6 Voice Translation Rules ..................................................................... 21.7 Shared Line........................................................................................ 31.8 Night Service...................................................................................... 31.9 After-Hours Setup .............................................................................. 31.10 Single Number Reach...................................................................... 31.11 Softkey Customization - SCCP ........................................................ 31.12 Softkey Customization - SIP ............................................................ 4

1.13 Octo-Line ......................................................................................... 41.14 Conference Resources .................................................................... 41.15 Transcoding Resources ................................................................... 41.16 B-ACD.............................................................................................. 51.17 Voice Hunt Groups........................................................................... 51.18 Ephone Hunt groups ........................................................................ 51.19 Dynamic Hunt groups ........................................................................ 51.20 Presence.......................................................................................... 51.21 Feature Access Codes..................................................................... 61.22 Call Park .......................................................................................... 61.23 Call Pickup ....................................................................................... 6

1.24 Music On Hold ................................................................................. 61.25 Speed Dials...................................................................................... 61.26 Intercom ........................................................................................... 61.27 Directory Services............................................................................ 71.28 Cisco Unity Express Enabling the AIM-CUE .................................. 71.29 Cisco Unity Express Initialization.................................................... 71.30 Cisco Unity Express Integration with CME ..................................... 71.31 Cisco Unity Express Phone Voicemail Options .............................. 71.32 Cisco Unity Express Advanced Features ....................................... 81.33 Cisco Unity Express Auto Attendant............................................... 81.34 Cisco Unity Express Live Record ................................................... 8

1.35 Cisco Unity Express Live Reply...................................................... 8


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Copyright 2009 Internetwork Expert INE.comiv

Cisco Unified Call Manager Express Solutions .................................. 91.1 CUCME Basic Configuration.............................................................. 91.2 Phone Registration & Number Assignment (SCCP) ........................ 121.3 SIP Phones ...................................................................................... 151.4 ISDN PRI ......................................................................................... 18

1.6 Voice Translation Rules ................................................................... 261.7 Shared Line...................................................................................... 291.8 Night Service.................................................................................... 301.9 After-Hours Setup ............................................................................ 321.10 Single Number Reach.................................................................... 331.11 Softkey Customization - SCCP ...................................................... 361.12 Softkey Customization - SIP .......................................................... 381.13 Octo-Line ....................................................................................... 391.14 Conference Resources .................................................................. 411.15 Transcoding Resources ................................................................. 431.16 B-ACD............................................................................................ 44

1.17 Voice Hunt Groups......................................................................... 461.18 Ephone Hunt groups ...................................................................... 471.19 Dynamic Hunt groups .................................................................... 481.20 Presence........................................................................................ 491.21 Feature Access Codes................................................................... 521.22 Call Park ........................................................................................ 531.23 Call Pickup ..................................................................................... 551.24 Music On Hold ............................................................................... 561.25 Speed Dials.................................................................................... 571.26 Intercom ......................................................................................... 581.27 Directory Services.......................................................................... 59

1.28 Cisco Unity Express Enabling the AIM-CUE ................................ 601.29 Cisco Unity Express Initialization.................................................. 631.30 Cisco Unity Express Integration with CME ................................... 671.31 Cisco Unity Express Phone Voicemail Options ............................ 771.32 Cisco Unity Express Advanced Features ..................................... 781.33 Cisco Unity Express Auto Attendant............................................. 801.34 Cisco Unity Express Live Record ................................................. 831.35 Cisco Unity Express Live Reply.................................................... 86


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Copyright 2009 Internetwork Expert INE.com1

Cisco Unified Communications

Manager Express

!Note

Many lab scenarios require previous labs to be properly configured. Therefore, itis recommended to save your config, and ensure you load the config beforeattempting further labs.

1.1 CUCME Basic Configuration

Configure the BR2 router (R3) as a Cisco Unified CommunicationsManager Express System (CUCME)

Ensure the voice interface is used as the source for the telephony service,and the digit 9 is used for secondary dialtone

Allow for a maximum of 4 e-phones and 32 ephone-dns

Set the time-zone to Pacific Standard/Daylight Time

Create a web enabled admin account. Username: admin Password: cisco

Ensure DNs and the TIME can be edited on the web interface

Set the time format to 24 hour and the date format to dd-mm-yy

Allow transfers and call-forwards to any destination

1.2 Phone Registration & Number Assignment (SCCP)

Configure R3 as a DHCP server for the phones connected to SW2

Only lease DHCP addresses between 177.3.11.100 and 177.3.11.110

Ensure the ports on SW2 are configured as mode access and cansupport a PC through the Phones switchport

Configure Br2Ph1 to register as a SCCP device

Assign phone numbers 3001 and 3005 to buttons 1 and 2 respectively

Assign labels to each button

Ensure the phone displays the fully qualified E.164 number at the top ofthe display

Ensure the calling name of each line is BR2Ph2

Assign Username: br2ph1and Password: ciscoto the phone

Assign a pin of 12345


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1.3 SIP Phones

Register Br2Ph2 as a sip phone with the router located at BR2

Allow for no more than 5 total SIP phone registrations

Allow for only 10 DNs

Apply a Calling Name in addition to a label next to the button

Apply number 3002 and 3003 to buttons 1 and 2 respectively


1.4 ISDN PRI

Configure R3 to set up an E1 port for ISDN PRI. Use only the first 3 slotsfor the PRI group, and CRC4/HDB3 as the framing/line-code settings

Ensure the calls from the PSTN can reach the IP Phones, and that thePSTN switch only sends 8-digits of subscriber ID to the router

Apply configuration to ensure caller-id information is transferred over thePRI

1.5 IOS Call Routing

Configure the following dialing patterns and route them using the localPSTN connection per the following:

o Emergency: 999

o Local: 9[1-8]XXXXXXX (8 digits)

o National: 90[1-8]XXXXXXXXX (10 digits)o International: 900+variable length

Strip caller ID information for national calls

Ensure Br2Ph1 can not call international numbers

1.6 Voice Translation Rules

Allow callers to reach 999 by dialing 123

Callers should be able to reach the PSTN number 4423221415555 by

dialing 7, then any three digits followed by 5555 All calls ending in 3XXX should be translated to only the last 4 digits

(Example: 1112223XXX Translated to: 3XXX)

Reject all calls beginning with 4567


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1.7 Shared Line

Configure Br2Ph2 as a SCCP phone

Configure line 3010 to be shared between button 6 on Br2Ph1 andBr2Ph3

(Note: This task requires registering a third IP Phone remotely (Hard orSoft Phone)

1.8 Night Service

Configure the night-service bell feature for Br2Ph1s primary line andconfigure so that DN 2 (3005) will be notified

Active the Night Service bell every day after 6pm and all day on Sat andSun

1.9 After-Hours Setup

Block the pattern 900 every day after 6pm till 9am but bypass the toll barfor a logged-in user for 15 minutes

Ensure Br2Ph2 is exempt from the After Hours call blocking

1.10 Single Number Reach Configure Single Number Reach for Br2Ph1

Configure the external call destination to 21415555 (PSTN Phone Line)

After a total of 20 seconds, redirect the call to Br2Ph2s primary line


1.11 Softkey Customization - SCCP

Configure a new ephone-template for use on Br2Ph1

Idle state keys should be NewCall Pickup Redial Login Hlog Seized state keys should be CallBack Cfwdall Endcall Gpickup

Connected state keys should be Endcall Hold Transfer Hlog


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1.12 Softkey Customization - SIP

Configure a new voice register template for user on Br2Ph2

Hold state keys should be Newcall Resume

Seized state keys should be Cfwdall Endcall Redial

Connected state keys should be Confrn Endcall Hold Trnsfer

1.13 Octo-Line

Enable the ability for Br2Ph1 to initiate an 8 Party conference using onedirectory number

Assign the number 3020 to Br2Ph1 button 3

After 5 calls to 3020, callers should hear a busy tone when calling

Br2Ph1 should still be able to make 3 more calls, even with five activecalls on button 3

1.14 Conference Resources

Configure R3 (BR2 router) to register the local NM-HDV DSP resourcesas a conference bridge with the local CallManager Express system

Source SCCP messages off the local voice VLAN subinterface

Configure the local CallManager IP address for the dsp-farm registration

Enable the dsp-farm service on the local voice port

Allow a maximum of 4 conference parties

1.15 Transcoding Resources

Configure R3 (BR2 router) to register the local NM-HDV DSP resourcesas a transcoder with the local CallManager Express system


Configure the local CallManager IP address for the dsp-farm registiration

Enable the dsp-farm service on the local voice port

Allow a maximum of 3 transcoding sessions

Configure the telephony-service settings to register the new dsp-farmunit and permit no more than 3 transcoding sessions


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1.16 B-ACD

Configure a B-ACD script using Auto-Attendant mode at the BR2 site

Use single pilot number 3000 and two hunt-groups 3100 and 3200covering BR2 IP Phone 1 and BR2 IP Phone 2 lines respectively

Allow dial by number when a user presses 3 and redirect the user toBR2 IP Phone 2 once a users presses 0 for an operator

1.17 Voice Hunt Groups

Create a hunt group with the pilot number 3000

Set lines 3001, 3002, and 3005 as members of the hunt group

Calls to the hunt group should be directed to all members (CallBlast/Broadcast)

After 20 seconds, calls should be disconnected

1.18 Ephone Hunt groups


Set lines 3001, and 3005 as the first members of the hunt group

Calls to the hunt group should be directed first to the number that is idlefor the longest period

Set the final destination to 3002

1.19 Dynamic Hunt groups


Allow lines 3001 and 3005 to join the group

Allow users to login and logout of the hunt group

Allow users to login by dialing a FAC (Feature Access Code) or bypressing a softkey

1.20 Presence

Enable the presence service

Allow the line 3003 on Br2Ph3 to be watched (SCCP Presentity) Allow the line 3002 on Br2Ph2 to be watched (SIP Presentity)

Configure Br2Ph1 as a watcher of 3003 using BLF Monitoring


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1.32 Cisco Unity Express Advanced Features

Allow the Br2Ph1 user to send broadcast messages via AvT(Administration via Telephone) application number 3555

Broadcast messages should trigger MWI lights

Create a new shared mailbox for both BR2 users with an extensionnumber 3000

Disable the CUE tutorial for both BR2 IP Phones

1.33 Cisco Unity Express Auto Attendant

Associate the CUE default Auto Attendant application with the extensionnumber 3600

Limit the number of concurrent calls to the AA script to 3

Edit the default business schedule to reflect the office working hours Mon-Fri 9:00am-5.00pm

Record and replace the default welcome greeting with the new one, whichsays Welcome to Branch 2 office

Set the operator number to BR2 IP Phone1s primary line number

1.34 Cisco Unity Express Live Record

Enable Live Record on both the CUCME and CUE systems

Set the Live Record Pilot to 4999

Set the recording reminder notification to repeat every 10 seconds

Ensure the reminder notification is set to the shortest possible duration

Enable the Live Record function for Br2Ph1

1.35 Cisco Unity Express Live Reply

Enable Live Reply on the CUE system Ensure that replies can not be sent to 3002

Replies should not be able to reach remote destinations

Use the extension variable to define to the calling-number-rule


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Cisco Unified Call Manager Express

Solutions

1.1 CUCME Basic Configuration

Configure the BR2 router (R3) as a Cisco Unified CommunicationsManager Express System (CUCME)

Ensure the voice interface is used as the source for the telephony service,and the digit 9 is used for secondary dialtone

Allow for a maximum of 4 e-phones and 32 ephone-dns

Set the time-zone to Pacific Standard/Daylight Time

Create a web enabled admin account. Username: admin Password: cisco

Ensure DNs and the TIME can be edited on the web interface

Set the time format to 24 hour and the date format to dd-mm-yy Allow transfers and call-forwards to any destination

Create a system message of: CUCME at BR2

Configuration

R3:!ntp master 1!telephony-servicemax-ephones 4

max-dn 32ip source-address 177.3.11.1time-zone 5time-format 24date-format dd-mm-yycall-forward pattern .Tweb admin system name admin password ciscodn-webedittime-webeditsystem message CUCME at BR2transfer-system full-consulttransfer-pattern .Tcreate cnf-files

secondary-dialtone 9

Explanation

To configure CUCME, the above basic configuration is necessary. To begin, therouter must either be configured to communicate with an NTP server, or beconfigured as an NTP master as shown above. The necessity for a time sourceis to ensure proper file tagging for the phone configurations.


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To enable call processing, and enable the Communications Manager Expressfunctionality, the telephony-servicecommand is issued, and the followingminimum configuration must be applied:

max-ephone sets the maximum number of phones allowed to register with the

CME system.max-dn sets the maximum number of configurable DNs or directory numbers.ip source-address sets the address which the system will use for VOIP.communication, the loopback address used above is the most reliable interfaceavailable on the router.create cnf-files- command issued to create a downloadable configuration filefor each phone registering with the system.

Verification

VORack01R3#show telephony-serviceCONFIG (Version=7.0(1))

=====================Version 7.0(1)Cisco Unified Communications Manager Expressip source-address 177.3.11.1 port 2000max-ephones 4max-dn 32max-conferences 8 gain -6dspfarm units 0dspfarm transcode sessions 0conference softwareprivacyno privacy-on-holdhunt-group report delay 1 hours

hunt-group logout DNDmax-redirect 5cnf-file location: system:cnf-file option: PER-PHONE-TYPEtime-format 24date-format dd-mm-yytimezone 5 Pacific Standard/Daylight Timesecondary-dialtone 9call-forward pattern .Ttransfer-pattern .Ttimeout interdigit 10timeout busy 10timeout ringing 180

timeout transfer-recall 0timeout ringin-callerid 8timeout night-service-bell 12caller-id name-only: enablesystem message CUCME at BR2

web admin system name admin password ciscoweb admin customer name Customeredit DN through Web: enabled.edit TIME through web: enabled.


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1.2 Phone Registration & Number Assignment (SCCP)

Configure R3 as a DHCP server for the phones connected to SW2

Only lease DHCP addresses between 177.3.11.100 and 177.3.11.110

Ensure the ports on SW2 are configured as mode access and cansupport a PC through the Phones switchport

Configure Br2Ph1 to register as a SCCP device

Assign phone numbers 3001 and 3005 to buttons 1 and 2 respectively

Assign labels to each button

Ensure the phone displays the fully qualified E.164 number at the top ofthe display

Ensure the calling name of each line is BR2Ph2


Assign a pin of 12345

Configuration

R3:

!! To ensure only a certain range of IPs are assigned, use the excluded-! address command to set ranges!ip dhcp excluded-address 177.3.11.1 177.3.11.99ip dhcp excluded-address 177.3.11.111 177.3.11.255!

ip dhcp pool VOICEnetwork 177.3.11.0 255.255.255.0default-router 177.3.11.1option 150 ip 177.3.11.1

!

SW2:

!interface range fa 0/1 2switchport mode accessswitchport access vlan 12switchport voice vlan 11!! To ensure STP delays are not experienced, use portfast!spanning-tree portfast!


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R3:!ephone-dn 1 dual-linenumber 3001label 3001description 21313001name Br2Ph1

!ephone-dn 2number 3005label 3005name Br2Ph1!ephone 1macusername br2ph1 password ciscopin 12345button 1:1 2:2restart!

Explanation

SCCP phones use the above method of configuration. Note the label, name, &description commands, which affect the appearance of the phone display, aswell as the calling name, which will be displayed to parties receiving calls fromthe phone. In addition, notice the maccommand, which will assign theautomatically registered MAC address to the phone, allowing for furtherconfiguration. In addition, ephone-dn is configured as a dual-line DN whichallows for conferencing.

Verification

VORack01SW2#show cdp nei fa 0/1 detail-------------------------Device ID: SEP001646807081Entry address(es):

IP address: 177.3.11.100Platform: Cisco IP Phone 7960, Capabilities: Host PhoneInterface: FastEthernet0/1, Port ID (outgoing port): Port 1

VORack01R3#show ip dhcp bindingBindings from all pools not associated with VRF:IP address Client-ID/ Lease expiration Type

Hardware address/User name

177.3.11.100 0100.1646.8070.81 Jun 03 2009 04:11 AM Automatic

177.3.11.101 0100.1646.6132.73 Jun 03 2009 04:12 AM Automatic

VORack01R3#show ephoneephone-1[0] Mac:0016.4680.7081 TCP socket:[1] activeLine:0 REGISTERED in SCCPver 11/9mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:5IP:177.3.11.100 49849 Telecaster 7960 keepalive 28 max_line 6button 1: dn 1 number 3001 CH1 IDLEbutton 2: dn 2 number 3005 CH1 IDLEPreferred Codec: g711ulawUsername: br2ph1 Password: cisco


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!Further Reading:http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmebasic.html#wp1028693


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1.3 SIP Phones

Register Br2Ph2 as a sip phone with the router located at BR2

Allow for no more than 5 total SIP phone registrations

Allow for only 10 DNs

Apply a Calling Name in addition to a label next to the button

Apply number 3002 and 3003 to buttons 1 and 2 respectively


Configuration

R3:!voice service voipallow-connections sip to sipsipregistrar server expires max 1200 min 300!voice register globalmode cmemax-pool 5max-dn 10source-address 177.3.11.1load 7960 P0S3-08-8-00tftp-path flash:upgradeauthenticate registercreate profile!tftp-server flash:P003-08-8-00.sbn

tftp-server flash:P003-08-8-00.bintftp-server flash:P0S3-08-8-00.sb2tftp-server flash:P0S3-08-8-00.loads!! Configure the DN!voice register dn 1number 3002name Br2Ph2label Br2 3002!! Configure Phones!voice register pool 1

!! Enter the mac address of your softphone!id mac ABCD.1234.5768type 7960number 1 dn 1dtmf-relay sip-notifyusername br2ph2 password ciscocodec g711ulaw


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Explanation

SIP endpoints require an IP-to-IP Gateway to be setup before calls will beprocessed or allowed for SIP phones. This is accomplished with the aboveconfiguration under voice service voipThe next step of the configuration issimilar to the Telephony Service configuration for SCCP endpoints. Finally, the

configuration of a voice register poolis required to register SIP phones with theCUCME system. The voice register pool is similar to the ephoneconfigurationfor SCCP. Note the id maccommand under the voice register pool 1whichmust be defined for a phone to register to that pool. If no mac address is defined,and a phone attempts to register a debug voice register erroroutput willappear as shown below:

Jun 2 15:33:04.779: VOICE_REG_POOL: Contact doesn't match any pools

Verification

VORack01R3#show voice register global

CONFIG [Version=7.1]========================

Version 7.1Mode is cmeMax-pool is 5Max-dn is 10Outbound-proxy is enabled and will use global configured valueSource-address is 177.3.11.1 port 5060Load 7960-40 is P0S3-08-8-00.loadsAuthenticate registerTime-format is 12Date-format is M/D/YTime-zone is 5Tftp path is flash:Generate text file is disabledTftp files are created, current syncinfo 0002234912431245OS79XX.TXT is not createdtimeout interdigit 10

!Note:

Notice the Authenticate register in the output above, which must be used withSIP Softphones.


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VORack01R3#show voice register pool 1Pool Tag 1Config:

Mac address is 0005.9A3C.7800Number list 1 : DN 1Proxy Ip address is 0.0.0.0DTMF Relay is enabled, sip-notify

Call Waiting is enabledDnD is disabledkeep-conference is enabledusername br2ph2 password ciscoTransport type is udpservice-control mechanism is not supportedregistration Call ID is OTZhNzcxMWU5ZWY0N2Y0MGY5YzhhZjRjMTRmNzdmYjE.active primary line is: 3002

contact IP address: 10.0.0.29 port 47630

Dialpeers created:

dial-peer voice 40001 voipdestination-pattern 3002session target ipv4:10.0.0.29:47630session protocol sipv2dtmf-relay sip-notifycodec g711ulaw bytes 160after-hours-exempt FALSE

Statistics:Active registrations : 1

Total SIP phones registered: 1Total Registration Statistics

Registration requests : 1

Registration success : 1Registration failed : 0unRegister requests : 0unRegister success : 0unRegister failed : 0

VORack01R3#show dial-peer voice summaryTAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET

STAT PORT40001 voip up up 3002 0 syst ipv4:10.0.0.29:4763020001 pots up up 3001$ 0 50/0/120002 pots up up 3005$ 0 50/0/2

!Note:

The above output is similar to show ephone, but with more detail. Notice thedial-peer created which is included in the output of the command. Auto createdSIP dial-peers are in the 40000 range, whereas SCCP peers are in the 20000range. The destination pattern is the phone itself for SIP, instead of a port.


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!Further Reading:http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmebasic.html#wp1073239

1.4 ISDN PRI

Configure R3 to set up an E1 port for ISDN PRI. Use only the first 3 slotsfor the PRI group, and CRC4/HDB3 as the framing/line-code settings

Ensure the calls from the PSTN can reach the IP Phones, and that thePSTN switch only sends 8-digits of subscriber ID to the router

Apply configuration to ensure caller-id information is transferred over thePRI

Configuration

R3:

!isdn switch-type primary-net5network-clock-participate wic 0!controller e1 1/0/0framing crc4linecode hdb3

pri-group timeslots 1-3!interface Serial 1/0/0:15isdn outgoing display-ie!! Handle all PSTN incoming calls using DID!dial-peer voice 1 potsincoming called-number .direct-inward-dialport 1/0/0:15!!

telephony-servicedialplan-pattern 1 21313... extension-length 4create cnf!


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Verification

VORack01R3(config-controller)#pri-group timeslots 1-3Jun 2 21:59:25.751: %LINK-3-UPDOWN: Interface ISDN-VOICE 1/0/0:15(1), changedstate to upJun 2 21:59:25.751: %LINK-3-UPDOWN: Interface ISDN-VOICE 1/0/0:15(2), changedstate to up

Jun 2 21:59:25.751: %LINK-3-UPDOWN: Interface ISDN-VOICE 1/0/0:15(3), changedstate to upJun 2 21:59:25.759: %ISDN-6-LAYER2UP: Layer 2 for Interface Se1/0/0:15, TEI 0changed to up

!Note:After applying thepri-groupcommand under the E1 controller, the router willautomatically create a serial interface. Note the interface numbering 1/0/0:15. E1PRI interfaces generate a :15 (1/0/0:15) addressing, while T1 PRI interfacesgenerate a :23 (1/0/0:23) addressing.

VORack01R3#show isdn statusGlobal ISDN Switchtype = primary-net5ISDN Serial1/0/0:15 interface

dsl 1, interface ISDN Switchtype = primary-net5Layer 1 Status:

ACTIVELayer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHEDLayer 3 Status:

0 Active Layer 3 Call(s)Active dsl 1 CCBs = 0The Free Channel Mask: 0x80000007Number of L2 Discards = 0, L2 Session ID = 0

Total Allocated ISDN CCBs = 0

VORack01R3#show isdn servicePRI Channel Statistics:ISDN Se1/0/0:15, Channel [1-31]

Configured Isdn Interface (dsl) 1Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1State : 0 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3

Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1State : 0 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2

!Note:To determine if an E1 or T1 PRI is UP and in the proper operational state, ensurethe isdn statusshows a Layer 2 status of: MULTIPLE_FRAME_ESTABLISHED.Under the isdn service output, you can see the timeslots or channels created andtheir state.

Next, place calls to BR2 DID numbers from the PSTN phone to verify that thedial-plan pattern has been configured correctly and the phones have registered:


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Alternate configuration examples:(Once applied, test calls once again to ensure the configuration is appliedcorrectly)

R3:

!! First remove the dialplan-pattern!telephony-serviceno dialplan-pattern 1

!!! Use of the secondary number entry!ephone-dn 1number 3001 secondary 21313001!

!! Use of a voice translation rule!voice translation-rule 10rule 1 /.*$3...$/ /2131\1/

!!voice translation-profile INCOMINGtranslate called 10

!!voip-incoming translation-profile INCOMING!

!Note:

Translation rules are explained in detail in the Voice Translation Rules Labbelow. However, notice the voip-incoming translation-profilecommand, whichapplies the translation rule globally to all calls entering the router. The rule couldinstead be applied to a dial-peer to match a specific incoming port or destinationpattern.

!Further Reading:

http://www.cisco.com/en/US/docs/ios/12_1/termserv/configuration/guide/dcdchant.html


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1.5 IOS Call Routing

Configure the following dialing patterns and route them using the localPSTN connection per the following:

o Emergency: 999

o Local: 9[1-8]XXXXXXX (8 digits)o National: 90[1-8]XXXXXXXXX (10 digits)

o International: 900+variable length

Strip caller ID information for national calls

Ensure Br2Ph1 can not call international numbers

Configuration

R3:!

dial-peer cor customname DEFAULTname PREMIUM!dial-peer cor list PT-DEFAULTmember DEFAULT!dial-peer cor list PT-PREMIUMmember PREMIUM!dial-peer cor list CSS-DEFAULTmember DEFAULT!dial-peer cor list CSS-PREMIUM

member PREMIUM!dial-peer voice 999 potsdestination-pattern 999port 1/0/0:15forward-digits 3corlist outgoing PT-DEFAULT!dial-peer voice 900 potsdestination-pattern 9[1-9].......$port 1/0/0:15corlist outgoing PT-DEFAULT!

dial-peer voice 901 potsdestination-pattern 90[1-9]Tport 1/0/0:15prefix 0corlist outgoing PT-DEFAULT!


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!dial-peer voice 902 potsdestination-pattern 900Tport 1/0/0:15prefix 00corlist outgoing PT-PREMIUM

!! Assign incoming COR lists to the Ephone DNs!ephone-dn 1corlist incoming CSS-DEFAULT!

Explanation

IOS Call Routing is based on Dial-Peers and the parameters that can beconfigured to affect them. In addition to features such as caller-id strip, dial-peerscan be blocked for incoming and outgoing calls, can be modified by translation-

rules and can point to voice ports, phones, gatekeepers, SIP endpoints/serversand other network devices.

The dial-peers created above are all POTS dial-peers instead of VOIP (There arealso Frame Relay and ATM dial peers, but they are not a topic for the CCIEVoice Lab). Pots dial-peers must point to a voice port, in this case (1/0/0:15)created by the E1 PRI configuration. The router matches digits entered on aphone, and then makes a routing decision based on the dial-peers configured.The closest match is chosen by the router and the call is then routed to theappropriate port, or network location.

Calls are broken into call legs (sections) which are used to determine a calls pathand the routing decisions made upon them. Each call leg is matched to a dial-peer in an inbound and outbound methodology. With this structure, a call must beable to match a dial-peer for both its incoming and outgoing call legs.

The criteria to match dial-peers is based on any of the following four attributes:

Incoming called-number (Called Number (DNIS) used to match inbound legs)Answer-address(Calling Number (ANI) used to match inbound legs)Destination-pattern (Calling Number for inbound, Called Number for outbound)Port (POTS Voice port the call will come in or go out)

** ANI Automatic Number Identification (The number of the phone making a call)** DNIS Dialed Number Identification Service (The number of the phone being called)


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The dial-peers configured in the task above make use of the destination-patternandportattributes. To match the number, a series of digits and regularexpressions are used. Each dial-peers relevant section is shown below with anexplanation of the digit matching used.

!

dial-peer voice 999 pots < The first line defines the dial-peer ID and typedestination-pattern 999 < The pattern matched here is strictly digitsport 1/0/0:15 < Defines the port that will be used for calls to 999forward-digits 3 < As 9 is recognized as an outside access code. The

forward-digits 3tells the router to forward the entire string of 999, instead ofsimply 99

!

corlist outgoing PT-DEFAULT < The COR restrictions being applied!dial-peer voice 900 pots

destination-pattern 9[1-9].......$ < Here, both digits and regularexpressions are used. The first 9 is the access code, which provides secondary-dialtone. Within the brackets [1-9] the 1-9 indicates a range, or any single digitbetween 1-9. Each . in the . represents any single digit between 1-9. Finallythe $ indicates the end of the string.

To summarize, this dial-peer will match a pattern beginning with 9 which will bestripped as it is recognized as the dialtone character, followed by a single digitbetween 1-9 (which eliminates 0) Finally, the number must be 7 digits longer, fora full length of 8 digits. The $ stops the digit matching. A dialed number of921415555 will match, whereas 902321415555 will not, as the 0 is not valid, andthe digit string is two digits too long.

!dial-peer voice 901 pots

destination-pattern 90[1-9]T< Here a similar structure is used from theprevious dial-peer, with the exception of the T which represents a variablelength string of any value. Any number beginning with 90 then 1-9 will bematched by this dial-peer.!prefix 0 < Because the 0 is before the [1-9] it will be stripped by the router asthe brackets represent the beginning of the string which will be sent out as thecalled number. The prefix command is used to reinsert the 0 before the rest of

the digit string.!dial-peer voice 902 pots

destination-pattern 900T < Once again the pattern ends with a T, and in thiscase represents international dialing, denoted by the 00 prefix (The UKinternational calling code)


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Verification

Place calls to the PSTN numbers from the phones at Br2 and ensure you canreach each pattern. Useful debug commands are: debug isdn q931, debugvoice dial-peer alland debug voice ccapi inout. In addition, you can use thecsim start command to perform call testing, as shown below to the PSTN. By

default, the command will return failure messages, however the call will in fact gothrough, as shown by the connected status.

VORack01R3#csim start 902321415555csim: called number = 902321415555, loop count = 1 ping count = 0

csim err csimDisconnected recvd DISC cid(9)csim: loop = 1, failed = 1csim: call attempted = 1, setup failed = 1, tone failed = 0Jun 3 15:12:37.583: ISDN Se1/0/0:15 Q931: Sending SETUP callref =0x0084 callID = 0x8005 switch = primary-net5 interface = UserJun 3 15:12:37.587: ISDN Se1/0/0:15 Q931: TX -> SETUP pd = 8 callref= 0x0084

Bearer Capability i = 0x8090A3Standard = CCITTTransfer Capability = SpeechTransfer Mode = CircuitTransfer Rate = 64 kbit/s

Channel ID i = 0xA98383Exclusive, Channel 3

Called Party Number i = 0x80, '02321415555'Plan:Unknown, Type:Unknown

Jun 3 15:12:37.611: ISDN Se1/0/0:15 Q931: RX


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1.6 Voice Translation Rules

Allow callers to reach 999 by dialing 123

Callers should be able to reach the PSTN number 4423221415555 bydialing 7, then any three digits followed by 5555

All calls ending in 3XXX should be translated to only the last 4 digits(Example: 1112223XXX Translated to: 3XXX)

Reject all calls beginning with 4567

Configuration

R3:!voice translation-rule 1rule 1 /123/ /999/!voice translation-profile 123-999translate called 1!voice translation-rule 2rule 1 /^7...$5555$/ /90044232141\1/!voice translation-profile TO_PSTNtranslate called 2!voice translation-rule 3rule 1 /.*$3...$/ /\1/!voice translation-profile 3XXXtranslate called 3!voice translation-rule 4rule 1 reject /^4567/!voice translation-profile REJECTtranslate called 4!dial-peer voice 123 potsdestination-pattern 123translation-profile outgoing 123-999port 1/0/0:15!dial-peer voice 456 potsdestination-pattern 4567translation-profile outgoing REJECT

port 1/0/0:15!dial-peer voice 7 potsdestination-pattern 7...Ttranslation-profile outgoing TO_PSTNport 1/0/0:15!


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Verification

To verify voice translation-rules, use the exec level command test voicetranslation-rule x yyy(Where X is the number of the rule, and yyy is the number

to simulate.VORack04R3#test voice translation-rule 1 123Matched with rule 1Original number: 123 Translated number: 911Original number type: none Translated number type: noneOriginal number plan: none Translated number plan: none

VORack04R3#test voice translation-rule 2 71485555Matched with rule 1Original number: 71485555 Translated number: 442321415555Original number type: none Translated number type: noneOriginal number plan: none Translated number plan: none



!Further Reading:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml


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1.7 Shared Line

Configure line 3010 to be shared between button 6 on Br2Ph1 andBr2Ph3

(Note: This task requires registering a third IP Phone remotely

Configuration

R3:

!ephone-dn 10label Shared Linenumber 3010!ephone 1button 6:10restart!ephone 2macdescription Br2Ph3button 6:10restart!

Explanation

An ephone-dn can be assigned to buttons on multiple phones creating a sharedline. This configuration can be useful for multiple reasons, such as the ability toanswer a call on multiple phones arriving at the same number. When a dn is inuse on one of the configured phones, other phones will show this status, and the

line will be in a state considered remote-in-use

Verification

VORack01R3#show ephoneephone-1[0] Mac:000C.29DB.3C63 TCP socket:[1] activeLine:0 REGISTEREDbutton 6: dn 10 number 3010 CH1 IDLE shared

ephone-2[1] Mac:0200.4C4F.4F50 TCP socket:[5] activeLine:0 REGISTEREDbutton 6: dn 10 number 3010 CH1 IDLE shared

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmebasic.html#wp1013228


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1.8 Night Service

Configure the night-service bell feature for Br2Ph1s primary line andconfigure so that Br2Ph3 will be notified

Activate the Night Service bell every day after 6pm and all day Saturday


Configuration

R3:!telephony-servicenight-service code *12345night-service day Mon 18:00 09:00night-service day Tue 18:00 09:00night-service day Wed 18:00 09:00night-service day Thu 18:00 09:00

night-service day Fri 18:00 09:00night-service day Sat 00:00 23:59

!! The line to monitor!ephone-dn 1night-service bell

!! The phone to notify!ephone 2night-service bell

Explanation

"#$ %&'#( )$*+&,$ -$.(/*$ .0012) -1* %1(&-&,.(&1%) (1 3$ )$%( (1 .%1(#$* 4$)(&%.(&1%

4/*&%' . 5*$64$-&%$4 (&7$ 5$*&148 "#$ 5$*&14 &) 4$-&%$4 /%4$* (#$ ($0$5#1%96)$*+&,$

5.*.7$($*): .%4 (#$ !"#$%&'()*"+( -(..,177.%4 ,.% 3$ .550&$4 (1 $&(#$* $5#1%$)1* $5#1%$64%)8

Verification

"1 +$*&-9 (#$ .31+$ ,1%-&'/*.(&1%: 50.,$ . ,.00 -*17 ;*?@@


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After dialing the night-service code *12345 (From either Ph1 or Ph3)

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmecover.html#wp1097582


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1.9 After-Hours Setup

Block the pattern 900 every day after 6pm until 9am but bypass the tollbar for a logged-in user for 15 minutes

Ensure Br2Ph2 is exempt from the After Hours call blocking

Configuration

R3:!telephony-serviceafter-hours day Sun 18:00 09:00after-hours day Mon 18:00 09:00after-hours day Tue 18:00 09:00after-hours day Wed 18:00 09:00after-hours day Thu 18:00 09:00after-hours day Fri 18:00 09:00after-hours day Sat 18:00 09:00

after-hours block pattern 1 900login timeout 15

!voice register pool 1after-hour exempt!

Explanation

The After-Hours call blocking feature is similar in configuration to the nightservice code. Note that you can exempt both SCCP and SIP phones, as well asindividual DNs and Dial Peers within the CME system. When a phone user logs

in using their pin code, the after-hours call blocking will be bypassed for theconfigured amount of time.

Verification

"1 +$*&-9 (#$ .31+$ ,1%-&'/*.(&1%: ,#.%'$ (#$ )9)($7 (&7$ (1 . (&7$ 2&(#&% (#$ .-($*6

#1/*) 5$*&14 1% (#$ *1/($*: /)&%' (#$ +./+0 '(%,177.%4 -*17 5*&+&0$'$4 $M$, 714$>$%.30$ 714$A N%,$ 91/ #.+$ ,#.%'$4 (#$ (&7$: .(($75( . ,.00 -*17 ;*


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1.10 Single Number Reach

Configure Single Number Reach for Br2Ph1

Configure the external call destination to 21415555 (PSTN Phone Line)

After a total of 20 seconds, redirect the call to Br2Ph2s primary line


Configuration

R3:!ephone-template 1softkeys idle Dnd Gpickup Pickup Mobilitysoftkeys connected Endcall Hold LiveRcd Mobility

!ephone-dn 1number 3001mobilitysnr 921415555 delay 5 timeout 15 cfwd-noan 3002

!ephone 1ephone-template 1restart!

Explanation

E&%'0$ D/73$* Q$.,# &) . %$2 -$.(/*$ &%(*14/,$4 &% JKL R8B8 "#$ -$.(/*$ .0012) -1*

. )&%'0$ 5#1%$ %/73$* (1 3$ /)$4 -1* 7/0(&50$ ,.00&%' 4$)(&%.(&1%) )/,# .) . /)$*)4$)P 5#1%$: ,$00 5#1%$: .%4 #17$ 5#1%$8 S% .44&(&1%: (#$ -$.(/*$ 1--$*) (#$ .3&0&(9 (1

)$.70$))09 (*.%)-$* . ,.00 3$(2$$% (#$ 4$)(&%.(&1%)8 "#$ 713&0&(9 )1-(P$9 &) /)$4 (1

(1''0$ (#$ -$.(/*$ 1% .%4 1-- -*17 (#$ ,1%-&'/*$4 S= =#1%$: &% .44&(&1% (1 5$*-1*7&%'

,.00 *1/(&%' -/%,(&1%) 4/*&%' .% .,(&+$ ,.008

Verification

"1 +$*&-9 (#$ .31+$ ,1%-&'/*.(&1%: $%)/*$ (#$ !"#$%$&' )"*&+,'.55$.*) 1% ;*


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D$M(: 50.,$ . ,.00 (1 ?@@B .%4 5&,P &( /58 D12 5/)# (#$ !"#$%$&' )"*&+,'.%4 )$0$,(

C),-. /0%% 1" !"#$%, 23"-,4.) )#12% 3$012U

V1/ ,.% .0)1 5/00 (#$ ,.00 3.,P (1 91/* 5#1%$ 1%,$ &( #.) 3$$% )$%( (1 (#$ 713&0$

4$)(&%.(&1%: 39 5*$))&%' (#$ *$)/7$ P$9U

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesnr.html


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1.11 Softkey Customization - SCCP

Configure a new ephone-template for use on Br2Ph1

Idle state keys should be NewCall Pickup Redial Login Hlog

Seized state keys should be CallBack Cfwdall Endcall Gpickup

Connected state keys should be Endcall Hold Trnsfer Hlog

Configuration

R3:!

$5#1%$6($750.($ B)1-(P$9) &40$ D$2J.00 =&,P/5 Q$4&.0 K13&0&(9 W1'&% X01'

)1-(P$9) )$&Y$4 J.00;.,P J-24.00 L%4,.00 Z5&,P/5

)1-(P$9) ,1%%$,($4 L%4,.00 X104 "*%)-$* X01'

[$5#1%$ B

$5#1%$6($750.($ B

*$)(.*(

[

Explanation

Softkeys are customizable, and are displayed above the bottom four buttons on aCisco 7960 IP phone. Each state that phone is in has different options forsoftkey customization, which are configured under the ephone-template. Notethat you can have multiple ephone-templates, and each phone can be set to use

a separate template. For a complete list of states and options for customization,follow the Further Reading link below.


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Verification

To verify the above task, simply restart the phone, and ensure the softkeylabeling appears correctly on your phone.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesoftk.html


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1.12 Softkey Customization - SIP

Configure a new voice register template for user on Br2Ph2

Hold state keys should be Newcall Resume

Seized state keys should be Cfwdall Endcall Redial

Connected state keys should be Confrn Endcall Hold Trnsfer

Configuration

R3:!voice register template 1softkeys connected Confrn Endcall Hold Trnsfersoftkeys hold Newcall Resume

softkeys seized Cfwdall Endcall Redial!voice register pool 1template 1!

Explanation

Softkeys are customizable, and are displayed above the bottom four buttons on aCisco 7960 IP phone. Each state that phone is in has different options forsoftkey customization, which are configured under the voice register template.Note that you can have multiple voice register templates, and each phone can be

set to use a separate template. For a complete list of states and options forcustomization, follow the Further Reading link below.

Verification

To verify the above task, simply restart the phone, and ensure the softkeylabeling appears correctly on your phone.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesoftk.html#wp1011759


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1.13 Octo-Line

Configure per the Communications Manager Express: BasicConfiguration & Phone Registration and Number Assignment (SCCP)scenario

Enable the ability for Br2Ph1 to initiate an 8 Party conference using onedirectory number

Assign the number 3020 to Br2Ph1 button 3

After 5 calls to 3020, callers should hear a busy tone when calling

Br2Ph1 should still be able to make 3 more calls, even with five activecalls on button 3

Configuration

R3:

!ephone-dn 10 octo-linenumber 3020

!ephone 1max-calls-per-button 7busy-trigger-per-button 5

button 3:10restart!

Explanation

"#$ 1,(160&%$ -$.(/*$ .0012) -1* \ ,#.%%$0) 5$* $5#1%$64%8 "#&) -$.(/*$ .0012) -1*'*$.($* ,.00 ,.5.,&(9: .) 2$00 .) (#$ )/551*( -1* 7/0(&50$ 5.*(9 ,1%-$*$%,$) 2&(#1/((#$ %$$4 -1* .44&(&1%.0 $5#1%$64%)8

Verification


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To verify the above configuration, use the following command: show ephone-dnsummary.As shown below, 8 channels have been created for ephone-dn 10(50/0/10)

VORack01R3#show ephone-dn summaryPORT CH DN STATE MWI_STATE CODEC VAD VTSP STATE

VPM STATE50/0/1 1 DOWN NONE - - -EFXS_INIT50/0/10 1 DOWN NONE - - -EFXS_INIT50/0/10 2 DOWN NONE - - -EFXS_INIT50/0/10 3 DOWN NONE - - -EFXS_INIT50/0/10 4 DOWN NONE - - -EFXS_INIT50/0/10 5 DOWN NONE - - -EFXS_INIT50/0/10 6 DOWN NONE - - -EFXS_INIT50/0/10 7 DOWN NONE - - -EFXS_INIT50/0/10 8 DOWN NONE - - -EFXS_INIT

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmebasic.html#wp1105668


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1.14 Conference Resources

Configure R3 (BR2 router) to register the local NM-HDV DSP resourcesas a conference bridge with the local CallManager Express system


Configure the local CallManager IP address for the dsp-farm registration Enable the dsp-farm service on the local voice port

Allow a maximum of 4 conference parties

Configuration

R3:!voice-card 0no dspfarmdsp services dspfarm!

sccp local FastEthernet0/0.11sccp ccm 177.3.11.1 identifier 1 version 7.0sccp!sccp ccm group 1associate ccm 1 priority 1associate profile 1 register R3_CONF

!dspfarm profile 1 conferencecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8codec g729r8

codec g729br8maximum sessions 2associate application SCCP

!telephony-servicesdspfarm units 2sdspfarm tag 1 R3_CONFconference hardware

!

Explanation

J1%-$*$%,&%' .0012) -1* 71*$ (#.% < 5.*(&$) (1 (.P$ 5.*( &% . ,.00: &% (#$ .31+$

,1%-&'/*.(&1%: %1(&,$ (#$ .001,.(&1% 1- *$)1/*,$) -*17 (#$ TE=O]QK (#.( .*$ /)$4 -1*

(#$ ,1%-$*$%,$ )$(/58 "#$ -&*)( )($5 &) 4$-&%&%' (#$ J.00 K.%.'$* 1* JKL )9)($7 (#.((#$ *$)1/*,$) 2&00 .))1,&.($ 2&(#8 "#&) &) .,,1750&)#$4 39 4$-&%&%' (#$ ,,7 &5

.44*$))8 D$M( 39 ,*$.(&%' . '*1/5: (#$ *1/($* 2&00 .))1,&.($ (#$ )5$,&-&$4 %.7$ 1- (#$

*$)1/*,$): .%4 -&%.009 2&(#&% (#$ 4)5-.*7 5*1-&0$: )/551*($4 ,14$,) .*$ 4$-&%$4: .)

2$00 .) (#$ 7.M&7/7 ,1%-$*$%,&%' )$))&1%)8 "1 ,1750$($ (#$ *$)1/*,$ .001,.(&1%:$%)/*$ 91/ )$( (#$ .55*15*&.($ )$((&%') /%4$* ($0$5#1%96)$*+&,$: .%4 4$-&%$ (#$

)4)5-.*7 (.'8


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Verification

To verify the above configuration, use the following commands:

VORack01R3#show sccpSCCP Admin State: UP

Gateway IP Address: 177.3.11.1Switchover Method: IMMEDIATE, Switchback Method: GUARD_TIMERSwitchback Guard Timer: 1200 sec, IP Precedence: 3Max Supported MTP sessions: 0User Masked Codec list: NoneTCP Link Status: CONNECTEDConferencing Oper State: ACTIVE - Cause Code: NONEActive Call Manager: 177.3.11.1, Port Number: 2000TCP Link Status: CONNECTEDCall Manager: 177.3.11.1, Port Number: 2000

Priority: 1, Version: 3.1 or Higher

VORack01R3#show dspfarmDSPFARM Configuration Information:Admin State: UP, Oper Status: ACTIVE - Cause code: NONETranscoding Sessions: 3(Avail: 3), Conferencing Sessions: 3(Avail: 3)Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout:600Connection check interval 600 Codec G729 VAD: ENABLED

!Further Reading:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008084fe1f.shtml


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1.16 B-ACD

Configure a B-ACD script using Auto-Attendant mode at the BR2 site

Use single pilot number 3000 and two hunt-groups 3100 and 3200covering BR2 IP Phone 1 and BR2 IP Phone 2 lines respectively

Allow dial by number when a user presses 3 and redirect the user toBR2 IP Phone 2 once a users presses 0 for an operator

Configuration

Perform the following steps to configure the B-ACD application for theCallManager Express System:

Step 1:Open the DocCD page corresponding to B-ACD script configuration: CiscoCallManager Express > B-ACD and Tcl Call-Handling Applications. Open the

Cisco Unified CME Basic Automatic Call Distribution and Auto-AttendantServicesection for reference.

Look for the example configuration named Cisco Unified CME B-ACD with OneAA: Example

Step 2:Copy and paste the sample configuration to the notepad. Edit the text to matchthe proper variables:

R3:

!

! Attach the application to a dial-peer

!

dial-peer voice 3000 potsservice aa

incoming called-number 21313000

port 1/0:15

!

ephone-hunt 1 longest-idle

pilot 3100

list 3001,3011

!

! Timeout every line after 3 seconds

!

timeout 3

!

ephone-hunt 2 longest-idle

pilot 3200

list 3002,3012

timeout 3

!


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! B-ACD application

application

!

! Adjust the script URL to the one you have in your flash

!

service queue flash:app-b-acd-2.1.2.2.tcl

param queue-len 10

param aa-hunt0 3002

param aa-hunt1 3100

param aa-hunt2 3200

param number-of-hunt-grps 2

param queue-manager-debugs 1

service aa flash:app-b-acd-aa-2.1.2.2.tcl

paramspace english location flash:

paramspace english index 1

paramspace english language en

param aa-pilot 21313000!

!Retry each group after 20 seconds

!

param call-retry-timer 20

!

! Present a second greeting to queued users every 30 seconds

!

param second-greeting-time 30

!

! Try to contract the internal groups for 300 seconds

!

param max-time-call-retry 300param max-time-vm-retry 2

!

! The name of the queue script

!

param service-name queue

param dial-by-extension-option 3

!

! When 300 seconds run off, call IP Phone 1

!

param voice-mail 3001

param number-of-hunt-grps 3

param handoff-string aa


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Verification

Call the DID number 2Y3Y3000 from the PSTN phone and make sure the IVRprompt responds. Press the 0, 1, 2 & 3 buttons and make sure the call getsrouted appropriately.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl.html

1.17 Voice Hunt Groups


Set lines 3001, 3002, and 3005 as members of the hunt group

Calls to the hunt group should be directed to all members (CallBlast/Broadcast)

After 20 seconds, calls should be disconnected

Configuration

R3:!voice hunt-group 1 parallelpilot 3000

list 3001, 3005, 3002timeout 20!

Explanation

Parallel hunt groups are similar to the shared line functionality available to SCCPphones. The primary difference is the support for SIP phones, and the formalconcept of configuring a hunt group.

Verification

To test the above configuration, perform the following steps: Dial 3000 from thePSTN phone, ensuring that the members of the hunt group ring.

!Further Reading:



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1.18 Ephone Hunt groups


Set lines 3001, and 3005 as the first members of the hunt group

Calls to the hunt group should be directed first to the number that is idle

for the longest period Set the final destination to 3002

Configuration

R3:!ephone-hunt 1 longest-idlepilot 3000list 3001, 3005final 3002!

Explanation

Hunt groups are used to hunt a call to multiple destinations. Multiple optionsexist for the call routing logic used to direct a call to a pilot number to a memberof the group. The following options can be figured after ephone-hunt 1[longest-idle | peer | sequential] Note that thepeeroption routes calls in around-robin fashion.

Verification

To test the above configuration, perform the following steps: Dial 3000 from

Br2Ph2, ensuring that the members of the hunt group ring. In addition, use thecommand show ephone-hunt.

!Further Reading:



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1.19 Dynamic Hunt groups


Allow lines 3001 and 3005 to join the group

Allow users to login and logout of the hunt group

Allow users to login by dialing a FAC (Feature Access Code) or bypressing a softkey

Configuration

R3:!telephony-servicefac standard!ephone-hunt 1 peerpilot 3000list *, *!ephone-dn 1number 3001ephone-hunt login!ephone-dn 3number 3005ephone-hunt login!ephone-template 1softkeys idle Newcall Redial Pickup Login Hlog

!ephone 1ephone-template 1!

Verification

To test the above configuration, perform the following steps: First, login bothmembers of the hunt group (ephone-dns 1 and 2) This can be accomplishedeither by using the softkey which is accessible on the phone, or by dialing thestandard fac (*3,XXXX Where XXXX is the extension of the hunt group pilot)*33000. Next, dial 3000 from Br2Ph2, ensuring that the members of the huntgroup ring. Log out both members, and test again.

!Further Reading:



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1.20 Presence

Enable the presence service

Allow the line 3003 on Br2Ph3 to be watched (SCCP Presentity)

Allow the line 3002 on Br2Ph2 to be watched (SIP Presentity)

Configure Br2Ph1 as a watcher of 3003 using BLF Monitoring

Configuration

!presencepresence call-list

!sip-uapresence enable

!

voice register dn 1allow watch!ephone-dn 5number 3003label 3003allow watch

!voice register dn 1number 3002allow watch!ephone 1blf-speed-dial 1 3002 label "Br2Ph2_Presence" device

blf-speed-dial 2 3003 label "Br2Ph3_Presence" devicerestart!

Explanation

Presence capablities have now been implemented into most of the Cisco UnifiedCommunications Products. The addition of presence to CUCME while not as indepth as CUCM offers the ablity to monitor devices and lines as a watcher. Thedevices or lines being monitored are known as Presentities.

A watcher phone makes use of blf-speed-dialswhich monitor the line or device

of the presentity. Once the presentity is considered to be in a busy state, thelamp next to the speed dial line will light, and a busy indication will be displayedon the screen, as shown in the verification section.


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Verification

VORack01R3#show presence globalPresence Global Configuration Information:=============================================Presence feature enable : TRUE

Presence allow external watchers : FALSEPresence max subscription allowed : 100Presence number of subscriptions : 2Presence allow external subscribe : FALSEPresence call list enable : TRUEPresence server IP address : 0.0.0.0Presence sccp blfsd retry interval : 60Presence sccp blfsd retry limit : 10Presence router mode : CME mode

Notice both phones (presentities) are now being monitored by Br2Ph1 (watcher)and are in the idle state as indicated by the phone icons on hook:


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Once a call is placed between both devices, both lamps will light next to the linesand the phone symbol shows off hook, indicating the presentities are busy:

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmepres.html


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1.21 Feature Access Codes

Define #456 as a custom redial FAC

Define #11 as a custom dial to voicemail FAC

Alias #5 to FAC #456 and #0 TO #11

Configuration

R3:

!telephony-servicevoicemail 3500fac custom redial #456fac custom voicemail #11fac custom alias 0 #5 to #456fac custom alias 1 #0 to #11

!

Explanation

Feature Access Codes (FAC) are by default enabled on CUCME systems. Inaddition to the standand FACs you can define custom codes as well as Aliaseswhich point to the cusom codes.

In the above configuration, you are defining codes for redialing as well as a codeto reach the configured voicemail system. (Note the voicemail system will not bereachable until you complete the applicable CUE labs)

Verification

You can verify the above configuration through the show run command inaddition to testing their functionality by dialing each code and its alias on aphone. (Note that the FAC will not show up under a show telephony-service)

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmefacs.html


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1.22 Call Park

Create one park slot that only Br2Ph1 can access

Create one park slot that is available to all users

Enable directed call park

Create a directed call park slot

Configure Br2ph1 to monitor park slot 7004

Configuration

R3:

!telephony-servicecall-park system application!ephone-dn 24number 7001park-slot timeout 10 limit 2 recall

description park slot for Br2Ph1!ephone-dn 25number 7555park-slot timeout 10 limit 2 recalldescription park slot for All Users

!ephone-dn 26number 7004park-slot directeddescription park-slot directed

!ephone 1button 4m26restart!

Explanation

Call Park slots are DNs that will park or hold a call after being activated by thepark softkey on a users phone. The advantage of call park versus simply placinga call on hold, is that once a call is parked, it can be retrieved from any phone onthe CUCME system. Call park slots are chosen based on closest match,therefore, park slot 7001 will be accessible to Br2Ph1, as its primary DN is 3001.

The CUCME matches DNs to park slots based on the last two digits on theconfigured DN number. If a match is not found for a DN, then the next availableslot is used (in this case 7555 for all users)

Directed call park allows a user to park a call at a specific slot. This isaccomplished using the Transfersoftkey, then dialing the park slot DN number.


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CUCME version 7.1 introduced more in depth features for call park whichessentially offer complete support for SIP phones and the ability to monitor apark slot using the button XmX command, where X represents the button and DNnumber.

Verification

VORack01R3#show ephone-dn park

DN 24 (7001) park-slot state FREENotify to () timeout 10 limit 2

Transfer back to originator

DN 25 (7555) park-slot state FREENotify to () timeout 10 limit 2

Transfer back to originator

Transfer back to originator retry 10 limit 2

DN 26 (7004) directed park-slot state FREENotify to () timeout 0 limit 0

To test basic call park: Place a call from from any phone at Br2, and attempt topark the call. Next, from another phone, dial the park slot DN number, and thecall should be picked up.

To test directed call park and line monitoring: Place a call from Br2Ph2 toBr2Ph3. Press the Transfersoftkey, and dial 7004 (the directed call park slot)Once the call is parked, you will be able to see that that park slot is in use on

Br2Ph1:

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmepark.html


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1.23 Call Pickup

Configure ephone-dn 1 in pickup-group 101

Configure ephone-dn 2 in pickup-group 202

Ensure the Pickupsoftkey is displayed on phones in the idle state

Configuration

R3:

!

ephone-dn 1pickup-group 101

ephone-dn 2pickup-group 202

!ephone-template 1softkeys idle Newcall Pickup Redial Mobility Login HLog

!ephone 1ephone-template 1restart!ephone 2ephone-template 1restart!

Explanation

The pickup softkey allows users to pickup calls from other DN destinations within

the CUCME system. The key can be pressed and then the user dials the DNwhere the call is located. This could be a park slot, ringing DN, or on hold DN.

Pickup groups allow for easy transfer in large call environmens, where multiplegroups or departments may be setup. Using the GPickup softkey, a user can dialthe group number to pickup a call for that group.

Verification

Place a call to a phone at Br2. Using the pickup softkey on another phone, youshould be able to dial the DN at which the call is being directed and pick up thecall. The DN can be in either a ringing, or on hold state. You can also use thepickup key for a call parked in a park slot.

!Further Reading:



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1.24 Music On Hold

Enable MOH

Source the audio file from local router flash

Stream the MOH to a multicast address of 239.1.1.1

Configuration

R3:!telephony-servicemoh music-on-hold.aumulticast moh 239.1.1.1 port 2000

!

Explanation

When a phone user places a call on hold, the party on hold will hear the audiobeing played from the moh source. In this case, the source is a .au file stored inthe routers flash. MOH can also be configured from a live source, which isdetailed in the further reading link below.

Verification

VORack01R3#dir flash:Directory of flash:/

76 -rw- 496521 music-on-hold.au

To verify the actual audio, place a call between Br2Ph1 and Br2ph2, place oneparty on hold, and ensure that you can hear the audio playing.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmemoh.html#wp1010372


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1.25 Speed Dials

Set up speed dials for Br2ph1 to the following numbers and label eachentry:

o 1 3002 (Br2Ph2)

o 2 900442321415555 (PSTN_LOCAL)o 3 90017755011111 (PSTN_HQ)

Configuration

R3:!ephone 1fastdial 1 3002 name Br2Ph2fastdial 2 900442321415555 name PSTN_LOCALfastdial 3 90017755011111 name PSTN_HQ

!

Explanation

Although referred to as Speed Dials, the configuration syntax uses fastdial. Eachentry allows the user to select the number on their phone, and the call will beplaced to that destination.

Verification

Select the services button on the IP phone, then select My Fast Dials. Select anentry, and ensure the call is placed correctly.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmespeed.html


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1.26 Intercom

Configure Standard Intercom between Br2Ph1 and Br2Ph3

Ensure the Barge-in option is enabled

Configure intercom that will allow the call even when the phone is busy

Configuration

R3:

!ephone-dn 16number 5001

whisper-intercom label "Whisper"!ephone-dn 17number 5002

whisper-intercom label "Whisper"

ephone-dn 18number 5003name "intercom"intercom 5002 barge-in

!ephone-dn 19name "intercom"number 5004intercom 5001 barge-in

!ephone 14:16 5:18

!ephone 2

4:17 5:19!ephone-dn 16number 5001

whisper-intercom label "Whisper"!ephone-dn 17number 5002

whisper-intercom label "Whisper"

Explanation

The intercom features in CUCME now offer two configuration options. The first isthe standard intercom, which will not work while a user is busy. The second is the

Whisper intercom, which will work even when a users phone is busy. Note thatwhisper intercom DNs can only call other intercom DNs configured in whispermode.

The intercom concept allows for a simple phone to phone conversation withouthaving to dial the other party. In addition to the simplicity of dialing, the intercomfeature will auto answer with speakerphone, similar to a building intecom system.


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Verification

From Br2Ph1 test the intercoms to Br2Ph3, and vice versa. Next, place a callfrom Br2Ph1 to an exernal destination, then ensure that the Whisper intercomstill works from Ph3.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinter.html

1.27 Directory Services

Configure per the Communications Manager Express: BasicConfiguration & Phone Registration and Number Assignment (SCCPPhones)scenarios

Create directory entries for Br2Ph1 and Br2Ph2

Configuration

R3:

!telephony-service

directory-entry 1 3001 name Br2Ph1directory-entry 2 3002 name Br2ph2!

Explanation

The directory service is enabled as soon as an entry has been added undertelephony-service. Each entry much be the in the format (Entry Identifier /Phone Number / name)

Verification

Press the directory button on one of the Br2 IP Phones, and ensure you canaccess and search the directory.

!Further Reading:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmedirs.html


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1.28 Cisco Unity Express Enabling the AIM-CUE

Enable the Service Engine in R3

Assign an IP and default gateway to the service module

Set a route to reach the Service Modules IP address

Test connectivity

Configuration

R3:!interface service-engine 0/0ip unnumbered fa 0/0.11no shut

!service-module ip address 177.3.11.254 255.255.255.0service-module ip default 177.3.11.1no shut!ip route 177.3.11.254 255.255.255.255 service-engine 0/0!

Explanation

Cisco Unity Express is the small scale messaging system in the Cisco UnifiedCommunications product suite. In the CCIE lab, the CUE system will be loadedon an AIM-CUE card, which is recognized by the router as a Service-Engine and

Service-Module. The Service-Engine is the addressable interface, which must beenabled to provide communication with the Service-Module. The address of theService-Engine can essentially be anything of your chosing, however it isrecommened to tie it to the voice interface (in this case fa 0.0/11) with the ip-unnumberedcommand.

After enabling the Service-Engine, you must define an IP and default-gateway forthe Service-Module, followed by an IP Route, which will instruct the router how todirect traffic to the CUE module itself. Note that the route points to the Service-Engine as this is the addressable interface the router recognizes. To verify theconfiguration of the route, you can perform both a ping to the Service-Modules IP

address, as well as perform a show ip routewhich will show all routesconfigured on the router. The route to the Service-Engine will show as a staticroute, as shown below.


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Verification

VORack01R3#show run | section Service-Engineinterface Service-Engine0/0ip unnumbered FastEthernet0/0.11service-module ip address 177.3.11.254 255.255.255.0service-module ip default-gateway 177.3.11.1

ip route 177.3.11.254 255.255.255.255 Service-Engine0/0

VORack01R3#ping 177.3.11.254

Type escape sequence to abort.Sending 5, 100-byte ICMP Echos to 177.3.11.254, timeout is 2 seconds:!!!!!Success rate is 100 percent (5/5), round-trip min/avg/max = 1/2/4 ms

VORack01R3#service-module service-Engine 0/0 statistics

Module Reset Statistics:CLI reset count = 0CLI reload count = 0Registration request timeout reset count = 0Error recovery timeout reset count = 0Module registration count = 1

VORack01R3#service-module service-Engine 0/0 statusService Module is Cisco Service-Engine0/0Service Module supports session via TTY line 194Service Module is in Steady stateService Module heartbeat-reset is enabledGetting status from the Service Module, please wait..Cisco Unity Express 7.0.3

CUE Running on AIMVORack01R3#show ip routeCodes: C - connected, S - static, R - RIP, M - mobile, B - BGP

D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter areaN1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2E1 - OSPF external type 1, E2 - OSPF external type 2i - IS-IS, su - IS-IS summary, L1 - IS-IS level-1, L2 - IS-IS

level-2ia - IS-IS inter area, * - candidate default, U - per-user

static routeo - ODR, P - periodic downloaded static route

Gateway of last resort is not set

177.0.0.0/24 is subnetted, 2 subnetsC 177.0.201.0 is directly connected, Serial0/0/1:0.1

O 177.0.101.0 [110/115] via 177.0.201.1, 19:05:39,Serial0/0/1:0.1177.1.0.0/16 is variably subnetted, 8 subnets, 2 masks

O 177.1.254.2/32 [110/116] via 177.0.201.1, 19:05:39,Serial0/0/1:0.1C 177.1.254.3/32 is directly connected, Loopback0O 177.1.254.1/32 [110/51] via 177.0.201.1, 19:05:39,Serial0/0/1:0.1O 177.1.19.0/24 [110/51] via 177.0.201.1, 19:05:39,Serial0/0/1:0.1


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O 177.1.254.254/32 [110/52] via 177.0.201.1, 19:05:40,Serial0/0/1:0.1O 177.1.10.0/24 [110/51] via 177.0.201.1, 19:05:40,Serial0/0/1:0.1O 177.1.11.0/24 [110/51] via 177.0.201.1, 19:05:40,Serial0/0/1:0.1O 177.1.12.0/24 [110/51] via 177.0.201.1, 19:05:40,

Serial0/0/1:0.1177.3.0.0/16 is variably subnetted, 3 subnets, 2 masks

S 177.3.11.254/32 is directly connected, Service-Engine0/0C 177.3.11.0/24 is directly connected, FastEthernet0/0.11C 177.3.12.0/24 is directly connected, FastEthernet0/0.12

10.0.0.0/24 is subnetted, 1 subnetsO 10.0.11.0 [110/53] via 177.0.201.1, 19:05:41, Serial0/0/1:0.1

177.253.0.0/24 is subnetted, 1 subnetsO 177.253.1.0 [110/52] via 177.0.201.1, 19:05:41, Serial0/0/1:0.1

!Further Reading:

http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a008037f2a9.shtml


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1.29 Cisco Unity Express Initialization

Reset the CUE Module to defaults

Perform the initialization configuration

Set the timezone to Great Britain

Set the administrator name to: admin

Set the password to: cisco

Configuration

VORack01R3#service-module service-Engine 0/0 sessionTrying 177.3.11.1, 2194 ... Opencue#cue# offline!!!WARNING!!!: If you are going offline to do a backup, it isrecommendedthat you save the current running configuration using the 'write'command,prior to going to the offline state.

Putting the system offline will terminate all end user sessions.

Are you sure you want to go offline[n]? : ycue(offline)#cue(offline)# restore factory default!!!WARNING!!!: This operation will cause all configuration and dataon the system to be erased. This operation is not re

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