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Application Note

Aastra MX-ONE (Ericsson MD110) version 3.2 SP1 using SIP trunk
to Cisco Unified Communications Manager 8.5
Jan 5 2011, Initial Version
Table of Contents
Introduction .............................................................................................................................................................................................................. 2
Network Topology .................................................................................................................................................................................................... 2
Limitations................................................................................................................................................................................................................ 3
System Components ................................................................................................................................................................................................. 4
Hardware Components ........................................................................................................................................................................................ 4
Software Requirements ....................................................................................................................................................................................... 4
Features .................................................................................................................................................................................................................... 5
Features Supported .............................................................................................................................................................................................. 5
Features Not Supported (See limitations) ............................................................................................................................................................ 5
Configuration ............................................................................................................................................................................................................ 6
Configuring the Aastra MX-One ......................................................................................................................................................................... 6
Configuring the Cisco Unified Communications Manager ............................................................................................................................... 12
Acronyms ............................................................................................................................................................................................................... 45

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EDCS# 967575 Rev# Initial Version

Introduction
This application note describes the necessary steps and configurations for connectivity between an Aastra MX-One (Ericsson MD110)
Version 3.2 SP1 and a Cisco Unified Communications Manager (CUCM) 8.5 with SIP trunk.
The network topology diagram (Figures 1) shows the test setup for end-to-end interoperability between Cisco Unified Communications
Manager Release 8.5 and the Aastra MX-One (Ericsson MD110) PBX using SIP trunk via IP connectivity. Features tested are basic call,
3-way (ad-hoc) conference, call transfer (attended and unattended), call forward (all, busy and no answer), hold/resume, and DTMF
interworking.

Network Topology

Cisco Unified
Communications Manager
Release 8.5
Cisco 3560 PoE
Ethernet Switch

Aastra MX-One
(Ericsson MD110)
Version 3.2 SP1
SIP Trunk
Digital Phone A2
x4200

Dial Plans
42XX
Dial-60XX

Digital Phone B2
x4201
Phone C1
x6000

Phone D1
x6001

Cisco 7960 SCCP


phones

Figure 1.

Phone C2
x6013

Phone D2
x6014

Cisco 7961 SIP phones

Basic Call Setup

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Limitations
Connected name and connected number cannot be presented during a call. As of the tested software version 3.2 SP1 the Aastra MXONE PBX does not support P-asserted-Id (PAI) nor Remote-party-Id (RPID) SIP headers.
During external call transfer name and number updates are not supported. PAI and RPID are not supported on Aastra MX-ONE PBX, as
of the tested software.
The Aastra MX-ONE PBX does not support external call forward unconditional (Follow-me across SIP trunk).
A call forward busy or no answer, initiated from a user on the Aastra MX-ONE PBX, includes a History-info SIP header which Cisco
Unified Communications Manager does not support. Due to this caveat centralized voice mail integration, using Cisco Unity, Unity
Connection or Unity Express, is not possible as of the tested software. Ciscos UM platforms require the SIP Diversion header for
correct functionality.
DTMF interwork limitation: Since the Aastra MX-ONE version 3.2 SP1 PBX does not support RFC2833 and Cisco Unified
Communications Manager does not support SIP INFO, DTMF interworking will not work between the two Systems.
When a user on the Cisco Unified Communications Manager side invokes call hold feature or there is media break during supplementary
services such as transfer or conference, CUCM sends a Re-INVITE with Audio-attribute inactive and when the call resumes, CUCM
sends Re-INVITE without SDP. Aastra MX-ONE PBX returns a 200 OK with SDP message with audio-attribute remaining in inactive
state, hence no voice path exist for the call. The workaround for this limitation is to enable send send-receive SDP in mid-call
INVITE checkbox in the SIP profile configuration and apply it to CUCM SIP trunk configuration for the Aastra MX-ONE PBX.
When a user on the Cisco Unified Communications Manager side invokes call hold feature, MOH was not heard on the Aastra MX-ONE
PBX phone. The workaround to get MOH streaming is to set CUCM service parameter Clusterwide Parameters (Service) -> Duplex
Streaming Enabled to True.
When a user on the Cisco Unified Communications Manager side invokes Early Attended call transfer, ringback tone was not heard on
the Aastra MX-ONE PBX phone. The workaround to get ringback tone is to set CUCM service parameter Clusterwide Parameters
(Service) -> Duplex Streaming Enabled to True.

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System Components
Hardware Components

Cisco MCS 7800 Unified Communications Manager Appliance


2 Cisco Unified IP phone 7960 configured as SCCP phones
2 Cisco Unified IP phone 7961 configured as SIP phones
Aastra MX-One (Ericsson MD110) PBX

Software Requirements
The following software is required:
Cisco Unified Communications Manager Release 8.5
Aastra MX-One Version 3.2 (See configuration section for details)

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Features
This section lists supported and unsupported features.
Features Supported
Basic calls
CLIP-Calling line (Number) identification presentation
CLIR-Calling line (Number) identification restriction
CNIP-Calling name identification presentation
CNIR-Calling name identification restriction
Consultation (Attended) transfer Local and Network/External (See Limitations section for details.)
Early Attended transfer Local and Network/External (See Limitations section for details.)
Call forward unconditional by join Local
Call forward unconditional by join Network/External (See Limitations section for details.)
Call forward busy by join Local
Call forward busy by join Network/External
Call forward no reply by join Local
Call forward no reply by join Network/External
Hold and resume (See Limitations section for details.)
3-way conference (ad-hoc)

Features Not Supported (See limitations)


COLP-Connected line (Number) identification presentation
COLR- Connected line (Number) identification restriction
CONP-Connected name identification presentation
CONR- Connected name identification restriction
Centralized Message center voicemail integration
DTMF interworking

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Configuration
This section contains configuration menus and commands and describes configuration sequences and tasks.
Configuring the Aastra MX-One
System Software Release
ROCAI
RODAI
ROEQI
RODDI
IPGKC

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System software release:


MDSH> ts_about;
====== MX-ONE Telephony System ======
Version: 3.2 SP1 build16
RPM Packages
============
Telephony Server 12.45.6 :
eri_sn-12.45.6-MR
Media Gateway File system 2.0 :
egx_rfs-2.0-1
Media Gateway 10.15 :
egx_sw-10.15-1
Media Gateway Classic 1.4_5 :
lsue_sw-1.4_5-1
Manager 8.48.1 :
eri_om-8.48.1-1

ROCAI:
Route category initiate
Set up internal characteristics for the route; for example, traffic direction, services, or bearer capabilities.
ROCAP:
Route category data print
MDSH> ROCAP:ROU=ALL;
ROUTE CATEGORY DATA
ROU SEL
TRM SERV
NODG DIST DISL TRAF SIG
BCAP
1 7110000000000010 5 3110000011 0 30 128 03151515 111110000031 111111
2 7130000000700040 5 3110000000 0 30 128 03151515 311110000030 001100
3 0110000000000010 5 3110000000 0 30 128 00151515 211100000031 111111
4 7130000000700040 5 3110000000 0 30 128 03151515 311110000030 001100
5 7130000000700040 5 3110000000 0 20 10 03151515 311110000030 001100
6 7130000000700040 5 3110000000 0 20 10 03151515 311110000030 001100
7 7110000000000010 5 3110000000 0 20 10 03151515 111110000031 001100
8 4130000000000010 4 2100000010 0 30 128 01151515 311110000050 111100
9 7110000000000010 4 3110000001 0 30 128 00151515 1111111000A1 001100
10 0110000000000010 5 3110000000 0 30 128 00151515 411110000000 001100
11 7110000000000010 5 3110000000 0 20 10 03151515 111110000011 111111
12 7110000000700010 5 3110000000 0 20 128 03151515 111110000011 111111
13 1120000100000010 5 2100000010 0 30 128 03151515 311100010050 001100
15 7110000000000010 4 3110000001 0 30 128 00151515 1111111000A1 001100
50 7110000000000010 5 3110000001 0 30 128 03151515 111111100091 001100
99 7110000000000010 5 3110000001 0 30 128 03151515 111111100091 001100
100 7110000000000010 4 3110000011 0 30 128 03151515 111100000031 111111
101 0110000000000010 4 2100000007 0 30 128 00151515 211100100030 001100
103 7110000000000010 4 3110000001 0 30 128 00151515 1111111000A1 101100
END
RODAI:
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Route data initiate


RODAP:
Route data print
MDSH> RODAP:ROU=15;
ROUTE DATA
ROU TYPE VARC
VARI
VARO
FILTER
15 TL66 H'00000000 H'00000000 H'00000111 NO
END
SIP Route:
> sip_route -set -route 15 -uristring0 "sip:?@172.20.214.254" -accept ALL
MDSH> sip_route -print -route 15
Route data for SIP destination
Route : 15
Protocol
= udp
Invite URI string 0 = sip:?@172.20.214.254
Invite URI string 1 =
Invite URI string 2 =
Invite URI string 3 =
Invite URI string 4 =
Invite URI string 5 =
Invite URI string 6 =
Invite URI string 7 =
From URI string 0 =
From URI string 1 =
From URI string 2 =
From URI string 3 =
From URI string 4 =
From URI string 5 =
From URI string 6 =
From URI string 7 =
Port
= default
Accepted calls = ALL
Priority
= 255
Match this
=
context B-party 0 =
context B-party 1 =
context B-party 2 =
context B-party 3 =
context B-party 4 =
context B-party 5 =
context B-party 6 =
context B-party 7 =
context A-party 0 =
context A-party 1 =
context A-party 2 =
context A-party 3 =
context A-party 4 =
context A-party 5 =
context A-party 6 =
context A-party 7 =
Authentication username =
Password
=
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Registration type = NO_REG


Trusted privacy domain = no
SOS a subscriber number =
SOS type of number
=
SOS destination number =

ROEQI:
Route equipment initiate
ROEDP:
Route equipment data print
>
MDSH> ROEDP:ROU=15,TRU=ALL;
ROUTE EQUIPMENT DATA
ROU TRU EQU
IP ADDRESS
15

001-1

172.20.174.30

SQU

INDDAT

CNTRL

H'000000000000

END

RODDI:
Route external destination date initiate
RODDP:
Route external destination data print
MDSH> RODDP:DEST=ALL;
EXTERNAL DESTINATION ROUTE DATA
DEST DRN ROU CHO CUST ADC
TRC SRT NUMACK PRE
10
100
160610000000025000600110000 0 1 0
11

50

160600000000025010600110100 0 1 0

12

170710000000025010700110100 0 1 0

13

140700000000025010400100100 0 1 0

20

000500000000025000000000000 0 1 0

235

101

30

15

170700000000025010700110100 0 1 0

33

170700000000025010700110100 0 1 0

35

100

40

100500000000025000000000000 0 1 0

41

15

170700000000025010700110100 0 1 0

43

140700000000025010400100100 0 1 0

44

000500000000025000000000000 1 1 0

150500000000025010500100100 0 1 0

150500000000025010500100100 0 1 0

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450

101

170700000000025010700110100 0 1 0

50

101

150500000000025010500100100 0 1 0

51

15

170700000000025010700110100 0 1 0

53

101

050500000000025010500100100 0 1 0

54

170700000000025010700110100 0 1 0

550

170700000000025010700110100 0 1 0

551

11

050500000000025000500110000 0 1 0

560

11

060600000000025000600110000 0 1 0

60

15

170700000000025010700110100 0 1 0

750

101

777

888

100

150500000000025010500100100 0 1 0

950

11

060600000000025000600110000 0 7 0

170700000000025010700110100 0 1 0
170700000000025010700110100 0 1 0

END
KSDDP:
Key system directory data print
MDSH> KSDDP:DIR=ALL;
KEY SYSTEM DIRECTORY DATA
DIR CUST EQU
CAT ADN ODN
4200
001-1-20-00 10 0
4201
001-1-20-01 10 0
4204
001-1-20-04 1
10 0
4205
001-1-20-05 1
10 0
4206
001-1-20-06 1
10 0
4207
001-1-20-07 1
10 0
4208
001-1-20-08 1
10 0
4209
001-1-20-09 1
10 0
4210
001-1-20-10 1
10 0
4211
001-1-20-11 1
10 0
4212
001-1-20-12 1
10 0
4213
001-1-20-13 1
10 0
4214
001-1-20-14 1
10 0
4215
001-1-20-15 1
10 0
4300
001-1-20-02 10 0
4301
001-1-20-03 1
10
0

CALALT TIMER

END

Calling name/connected name restrictions


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> MDSH> KSCAP:DIR=4200&4201;


KEY SYSTEM CATEGORY PRINT
DIR TRAF
SERV
CDIV
ROC ITYPE TRM ADC
LANG
BSEC
4200 00151515 0202720500 011151111 000001 20 0 00100013010000 F
0
4201 03151515 0211120700 111151111 720004 20 0 00100013010000 F
0
END
To configure Calling/Connected Name and Number Restricted, use the following command:
<KSCAC:DIR=4200&&4201,ADC=00010013010000;
To configure Calling/Connected Name and Number Allowed, use the following command:
<KSCAC:DIR=4200&&4201,ADC=00100013010000;
Call diversion on busy/no reply
MDSH> CDIDP:DIR=ALL;
NOT ACCEPTED
DIV
NOT ASSIGNED
To enable diversion on busy/no reply, use the following command:
MDSH> CDINI:DIR=4200,DIV=4201; //Call diversion individual number initiate
EXECUTED
To show diversion settings:
MDSH> CDIDP:DIR=ALL;
CALL DIVERSION INDIVIDUAL DATA
DIR
DIV
4200
4201
END
To disable diversion on busy/no reply, use the following command:
MDSH> CDINE:DIR=4200; //Call diversion individual number end
EXECUTED
Codec preference list
IPGKC:
MDSH> IPGKC:LIM=1,CODECS=C-A-B-9-0-1-2-3-4-5-6-7-8;
EXECUTED
MDSH> IPGKP:LIM=1;
GATEKEEPER DATA
LIM TMLOAD CODECS
1
0
CAB9012345678

GKID
LIM1

END

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Configuring the Cisco Unified Communications Manager


1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.

Cisco Unified Communications Manager Version


Region Configuration
SIP profile Configuration
SIP Trunk Configuration
Media Resource Group List configuration
Media Resource Group configuration
Route Pattern Configuration
Calling Name and Number Restrictions
Connected Name and Number Restrictions
Cisco IP Phone 7960 SCCP Configuration
Cisco IP Phone 7961 SIP Configuration
Service parameters Configuration

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Cisco Unified Communications Manager Version

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Configuring the Region


Navigation Path: System -> Region

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SIP profile Configuration (1 of 2)


Navigation Path: Device -> Device settings-> SIP Profile

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SIP Profile Configuration (2 of 2)

In order to get bidirectional voice path


after call Hold/Resume or media break
during supplementary Services, the
send send-receive SDP in mid-call
INVITE needs to be checked.

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SIP Trunk Configuration (1 of 3)


Navigation Path: Device -> Trunk

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Trunk Configuration (2 of 3)
Navigation Path: Device -> Trunk

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Trunk Configuration (3 of 3)
Navigation Path: Device -> Trunk

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Media Resource Group List configuration


Navigation Path: Media Resources -> Media Resource Group List

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Media Resource Group configuration


Navigation Path: Media Resources -> Media Resource Group

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Route Pattern Configuration 42XX (1 of 2)


Navigation Path: Call Routing -> Route/Hunt -> Route Pattern

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Route Pattern Configuration 42XX (2 of 2)


Navigation Path: Call Routing -> Route/Hunt -> Route Pattern

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Calling Name and Number Restriction Configuration


Navigation Path: Device -> Trunk

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Connected Name and Number Restriction Configuration


Navigation Path: Device -> Trunk

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Cisco IP Phone 7960 SCCP Configuration (1 of 4)


Navigation Path: Device -> Phone

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Cisco IP Phone 7960 SCCP Configuration (2 of 4)


Navigation Path: Device -> Phone

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Cisco IP Phone 7960 SCCP Configuration (3 of 4)


Navigation Path: Device -> Phone

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Cisco IP Phone 7960 SCCP Configuration (4 of 4)


Navigation Path: Device -> Phone

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Cisco IP Phone 7960 SCCP DN Configuration (1 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7960 SCCP DN Configuration (2 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7960 SCCP DN Configuration (3 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7960 SCCP DN Configuration (4 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7961 SIP Configuration (1 of 6)


Navigation Path: Device -> Phone

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Cisco IP Phone 7961 SIP Configuration (2 of 6)


Navigation Path: Device -> Phone

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Cisco IP Phone 7961 SIP Configuration (3 of 6)


Navigation Path: Device -> Phone

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Cisco IP Phone 7961 SIP Configuration (4 of 6)


Navigation Path: Device -> Phone

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Cisco IP Phone 7961 SIP Configuration (5 of 6)


Navigation Path: Device -> Phone

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Cisco IP Phone 7961 SIP Configuration (6 of 6)


Navigation Path: Device -> Phone

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Cisco IP Phone 7961 SIP DN Configuration (1 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7961 SIP DN Configuration (2 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7961 SIP DN Configuration (3 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Cisco IP Phone 7961 SIP DN Configuration (4 of 4)


Navigation Path: Device -> Phone -> Modify/Add Button

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Configuring Service Parameter Duplex Streaming


Navigation Path: System -> Service parameters -> Cisco CallManager (Active)

To get MOH streaming on


Aastra MX-One station, set
CUCM service parameter

Clusterwide Parameters
(Service) -> Duplex
Streaming Enabled to True.
This setting also used for
ringback tone to be heard on
the Aastra MX-One station
when Early Attended
transfer is performed from
CUCM side.

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Acronyms

Acronym

Definitions

CUCM

Cisco Unified Communications Manager

CFB

Call Forwarding on Busy

CFNR

Call Forwarding No Reply

CFU

Call Forwarding Unconditional

CLIP

Calling Line (Number) Identification Presentation

CLIR

Calling Line (Number) Identification Restriction

CNIP

Calling Name Identification Presentation

CNIR

Calling Name Identification Restriction

COLP

Connected Line (Number) Identification Presentation

COLR

Connected Line (Number) Identification Restriction

CONP

Connected Name Identification Presentation

CONR

Connected Name Identification Restriction

CT

Call Transfer

MWI

Message Waiting Indicator

MOH

Music On Hold

PSTN

Public Switched Telephone Network

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Important Information
THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE
WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO
BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE
FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.
IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR
INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA
ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN
ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.

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Corporate
Headquarters

European
Headquarters

Americas
Headquarters

Asia Pacific
Headquarters

Cisco Systems, Inc.


170 West Tasman Drive
San Jose, CA 95134-1706
USA
www.cisco.com
Tel: 408 526-4000
800 553-NETS (6387)
Fax: 408 526-4100

Cisco Systems International


BV
HArlerbergpark
HArlerbergweg 13-19
1101 CH Amsterdam
The Netherlands
www-europe.cisco.com
Tel: 31 0 20 357 1000
Fax: 31 0 20 357 1100

Cisco Systems, Inc.


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San Jose, CA 95134-1706
USA
www.cisco.com
Tel: 408 526-7660
Fax: 408 527-0883

Cisco Systems, Inc.


Capital Tower
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#22-01 to #29-01
Singapore 068912
www.cisco.com
Tel: +65 317 7777
Fax: +65 317 7799

Cisco Systems has more than 200 offices in the following countries and regions. Addresses, phone numbers, and fax numbers are listed on
the Cisco Web site at www.cisco.com/go/offices.

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2008 Cisco Systems, Inc. All rights reserved.
CCENT, Cisco Lumin, Cisco Nexus, Cisco TelePresence, the Cisco logo and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Ciso
Store and Changing the Way We Work, Live, Play, and Learn are service marks of Cisco Systems, Inc.; and Access Registrar, Aironet, BPX, Catalyst,
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All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
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2009 Cisco Systems, Inc. All rights reserved.


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