aastra mx-one (ericsson md110) version 3.2 sp1 using sip ... · aastra mx-one pbx returns a 200 ok...

© 2009 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

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

© 2009 Cisco Systems, Inc. All rights reserved.

Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com of 47

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



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.



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)



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



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



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:



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 =







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 =



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



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



> 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



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



Cisco Unified Communications Manager Version



Configuring the Region

Navigation Path: System -> Region



SIP profile Configuration (1 of 2)

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



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.



SIP Trunk Configuration (1 of 3)

Navigation Path: Device -> Trunk



Trunk Configuration (2 of 3)




Trunk Configuration (3 of 3)




Media Resource Group List configuration

Navigation Path: Media Resources -> Media Resource Group List



Media Resource Group configuration

Navigation Path: Media Resources -> Media Resource Group



Route Pattern Configuration 42XX (1 of 2)

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



Route Pattern Configuration 42XX (2 of 2)

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



Calling Name and Number Restriction Configuration




Connected Name and Number Restriction Configuration




Cisco IP Phone 7960 SCCP Configuration (1 of 4)

Navigation Path: Device -> Phone



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

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



Cisco IP Phone 7961 SIP Configuration (1 of 6)




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




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.



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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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

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aastra mx-one (ericsson md110) version 3.2 sp1 using sip ... · aastra mx-one pbx returns a 200 ok...

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