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Page 1 of 47 EDCS# 967575 Rev# Initial Version
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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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
Digital Phone B2
x4201
Digital Phone A2
x4200
Aastra MX-One
(Ericsson MD110)
Version 3.2 SP1
Dial-60XX
Cisco Unified
Communications Manager
Release 8.5
SIP Trunk
Cisco 3560 PoE
Ethernet Switch
Cisco 7960 SCCP
phones
Dial Plans
42XX
Cisco 7961 SIP phones
Phone D1
x6001
Phone C2
x6013
Phone D2
x6014Phone C1
x6000
Figure 1. 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 MX-
ONE 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. Cisco’s 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:[email protected]" -accept ALL
MDSH> sip_route -print -route 15
Route data for SIP destination
Route : 15
Protocol = udp
Invite URI string 0 = sip:[email protected]
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 SQU INDDAT CNTRL
15 001-1 172.20.174.30 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 3 170710000000025010700110100 0 1 0
13 1 140700000000025010400100100 0 1 0
20 8 000500000000025000000000000 0 1 0
235 101 150500000000025010500100100 0 1 0
30 15 170700000000025010700110100 0 1 0
33 9 170700000000025010700110100 0 1 0
35 100 150500000000025010500100100 0 1 0
40 8 100500000000025000000000000 0 1 0
41 15 170700000000025010700110100 0 1 0
43 1 140700000000025010400100100 0 1 0
44 8 000500000000025000000000000 1 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 9 170700000000025010700110100 0 1 0
550 9 170700000000025010700110100 0 1 0
551 11 050500000000025000500110000 0 1 0
560 11 060600000000025000600110000 0 1 0
60 15 170700000000025010700110100 0 1 0
750 101 170700000000025010700110100 0 1 0
777 3 170700000000025010700110100 0 1 0
888 100 150500000000025010500100100 0 1 0
950 11 060600000000025000600110000 0 7 0
END
KSDDP:
Key system directory data print
MDSH> KSDDP:DIR=ALL;
KEY SYSTEM DIRECTORY DATA
DIR CUST EQU CAT ADN ODN CALALT TIMER
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
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 GKID
1 0 CAB9012345678 LIM1
END
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Configuring the Cisco Unified Communications Manager
1. Cisco Unified Communications Manager Version
2. Region Configuration
3. SIP profile Configuration
4. SIP Trunk Configuration
5. Media Resource Group List configuration
6. Media Resource Group configuration
7. Route Pattern Configuration
8. Calling Name and Number Restrictions
9. Connected Name and Number Restrictions
10. Cisco IP Phone 7960 SCCP Configuration
11. Cisco IP Phone 7961 SIP Configuration
12. 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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