Interactive Intelligence Messaging Interaction Center 2.4

Interactive Intelligence
,

Inc.

7601 Interactive Way

Indianapolis, Indiana 46278

Telephone/Fax (317) 872
-
3000

www.ININ.com





Interactive Intelligence Messaging Interaction Center 2.4

PBX Configuration Note:

Mitel 3300 with AudioCodes Mediant 1000/2000 using T1 Q.SIG

Technical Reference

By AudioCodes


READ THIS BEFORE YOU PROCEED

This document is for informational purposes on
ly and is provided “AS IS.” Interactive Intelligence, its partners and vendors
cannot verify the accuracy of this information and take no responsibility for the content of this document.
INTERACTIVE
INTELLIGENCE, ITS PARTNERS AND VENDORS MAKE NO WARRANTIES
, EXPRESS, IMPLIED OR STATUTORY,
AS TO THE INFORMATION IN THIS DOCUMENT.

Customer Interaction
Center
®

Vonexus

Enterprise Interaction Center
®

Document
Version

1.0

Last updated
:

05
-
19
-
2008

Content

This document describes the config
uration required to setup
A
vaya S8
3
00
and AudioCodes
Mediant 2000 using
T1 CAS In
-
band DTMF Tones
as the telephony signaling protocol. It also
contains the results of the interoperability testing of Interactive Intelligence Messaging
Interaction Center 2.4 (MIC) based on this setup.






i

Copyright and Trademark Information

Copyright ©1994
–

2008 Interactive Intelligence Inc. All rights reserved.
Interactive Intelligence®,
Vonexus®, Interaction Center Platform®, Communité®,
Vonexus

Enterprise Interaction Center®,
Interactive Intelligen
ce Customer Interaction Center®, e
-
FAQ®, e
-
FAQ Knowledge Manager, Interaction
Dialer®, Interaction Director®, Interaction Gateway, Interaction Marquee,

Interaction Mobile Office,
Interaction Optimizer, Interaction Recorder®,
Interaction Screen Recorder
, In
teraction SIP Proxy,
Interaction Supervisor, Interaction Tracker®, Mobilité®, SIP Interaction Media Server, Vocalité®,
Interaction Administrator®, Interaction Attendant®, Interaction Client®, Interaction Designer®,
Interaction Fax Viewer, Interaction FAQ,
Interaction Melder, Interaction Scripter®, Interaction
EasyScripter, Interaction Server, Interaction Voic
e
-
mail

Player, iRecord, Wireless Interaction Client,
Interactive Intelligence Live Conference, Vonexus Live Conference, icNotify, InteractiveLease, the

Vonexus
logo design® and the Interactive Intelligence “Spirograph” logo design® are all trademarks or registered
trademarks of Interactive Intelligence Inc.

veryPDF
is

Copyright © 2000
-
2005 by veryPDF, Inc.

Other brand and/or product names referenced in t
his
document are the trademarks or registered trademarks of their respective companies.

NOTICE

This product contains Information and/or data of Telcordia Technologies, Inc. (Telcordia) licensed to be
included herein. Recipient acknowledges and agrees that
(1) TELCORDIA AND ITS AFFILIATES MAKE NO
REPRESENTATIONS, EXTEND NO WARRANTIES OF ANY KIND, EXPRESSED OR IMPLIED, AND ASSUME NO
RESPONSIBILITY OR LIABILITY WHATSOEVER WITH RESPECT TO THE USE, SUFFICIENCY OR ACCURACY
OF THE PRODUCT, (2) RECIPIENT SHALL MAKE

NO CLAIM AGAINST TELCORDIA OR ANY OF ITS
AFFILIATES WITH RESPECT TO THE PRODUCT, AND WAIVES ALL CLAIMS AGAINST TELCORDIA OR ANY
OF ITS AFFILIATES WITH RESPECT TO THE PRODUCT, (3) IN NO EVENT SHALL TELCORDIA OR ANY OF
ITS AFFILIATES BE LIABLE FOR ANY DAMAG
ES, INCLUDING ANY LOST PROFITS OR OTHER INCIDENTAL
OR CONSEQUENTIAL DAMAGES RELATING TO THE PRODUCT, AND, (4) THIS AGREEMENT SHALL BE
ENFORCEABLE BY TELCORDIA.

DISCLAIMER

INTERACTIVE INTELLIG
ENCE (INTERACTIVE) H
AS NO RESPONSIBILITY

UNDER WARRANTY,
INDEMNIF
ICATION OR OTHERWISE
, FOR MODIFICATION O
R CUSTOMIZATION OF A
NY INTERACTIVE
SOFTWARE BY INTERACT
IVE, CUSTOMER OR ANY

THIRD PARTY EVEN IF
SUCH CUSTOMIZATION A
ND/OR
MODIFICATION IS DONE

USING INTERACTIVE TO
OLS, TRAINING OR MET
HODS DOCUMENTED BY
INTERACTIVE.


Interactive Intelligence Inc.

7601 Interactive Way

Indianapolis, Indiana 46278

Telephone/Fax (317) 872
-
3000

www.ININ.com

Interaction Center Platform Statement

This document describes Interaction Center (IC) features tha
t may not be available in your IC
product. Several products are based on the IC platform, and some features are disabled in
some products.

Th
ese

products are based on the IC platform:



Customer Interaction Center (CIC)



Vonexus

Enterprise Interaction Cente
r (
Vonexus

EIC, or EIC)



Message Interaction Center (MIC)

While all of these products share a common feature set, this document is intended for use with
all IC products, and some of the described features may not be available in your product.



ii

Table of Con
tents

Who should read this document

................................
................................
..................

1

Technical Support

................................
................................
................................
............

1

Chapter 1:
General Information

................................
................................
.................

2

Components

................................
................................
................................
.....................

2

PBX or IP
-
PBX

................................
................................
................................
................................
......................

2

VoIP Gateway
................................
................................
................................
................................
.......................

2

Interactive Intelligence Messaging Interaction Center

................................
................................
........

2

Prerequisites

................................
................................
................................
....................

2

Gateway Prerequisites

................................
................................
................................
................................
......

2

PBX Prerequisites
................................
................................
................................
................................
................

2

Cabling Requirements
................................
................................
................................
................................
.......

3

Summary and Limitations

................................
................................
...............................

3

Chapter 2:

Gateway Setup

................................
................................
..........................

4

Step 1: SIP Environment Setup

................................
................................
.....................

4

Step 2: Routing Setup
................................
................................
................................
.....

5

Step 3: Coder Setup
................................
................................
................................
........

6

Step 4: Digit Collection Setup

................................
................................
........................

7

Step 5: Supplementary Services Setting
................................
................................
.......

8

Step 6: Trunk Group Setup

................................
................................
............................

9

Step 7: TD
M Bus Setting
................................
................................
...............................

10

Step 8: Trunk Setting Setup
................................
................................
.........................

11

Step 9: Fax Setup
................................
................................
................................
..........

12

Step 10: Voicemail Setup

................................
................................
.............................

13

Step 11: General Setup

................................
................................
................................

14

Step 12: General Setup (Cont.)

................................
................................
...................

15

Step 13: Reset Mediant 2000

................................
................................
.......................

16

Configuration Files
................................
................................
................................
................................
............

16

................................
................................
................................
................................
...............

16

Chapter 3: PBX Setup

................................
................................
................................
...

17

Step 1: Ensure PBX is equipped with T1 PRI Module to Support T1 QSIG

...............

17

Fail
-
Over Configuration

................................
................................
................................
................................
..

36

Tested Phones
................................
................................
................................
................................
....................

36

Other Comments
................................
................................
................................
................................
...............

36



iii

Chapter 4:

Messaging Interaction Center 2.4 Validation Test Matrix

...........

37

Testing the Core Feature Set
................................
................................
........................

37

Detailed Descript
ion of Limitations

................................
................................
..............

38

Chapter 5:

Troubleshooting

................................
................................
......................

39

Appendix

................................
................................
................................
..........................

41

Dial Pilot Number and Mailbox Login

................................
................................
...........

41

Navigate Mailbox using Mobile Office
................................
................................
...........

41

Navigate Mailbox using Telephony User Interface (
TUI)

................................
............

41

Dial User Extension and Leave Voicemail

................................
................................
....

41

From an Internal Extension

................................
................................
................................
..........................

41

From an External Phone
................................
................................
................................
................................
.

41

Dial Auto Attendant (AA)

................................
................................
..............................

42

Call Transfer by Dial By Name
................................
................................
......................

42

Called Party is Busy

................................
................................
................................
................................
.........

42

Called Party does not Answer

................................
................................
................................
......................

42

Voicemail Button

................................
................................
................................
............

42

Testing Fax Features

................................
................................
................................
.....

42

Message Waiting Indicator (MWI)

................................
................................
................

43

Appendix A: Chang
e Log

................................
................................
..............................

44

Appendix B: Acronyms Used in This Document

................................
.....................

45



1

Who should read this document

This document is intended for Systems Integrators with significant telepho
ny knowledge.


Technical Support

The information contained within this document has been provided by
Interactive Intelligence, its

partners or equipment manufacturers and is provided AS IS. This document contains information
about how to modify the configu
ration of your PBX or VoIP gateway. Improper configuration may
result in the loss of service of the PBX or gateway.
Interactive Intelligence

is unable to provide support
or assistance with the configuration or troubleshooting of components described within
.
Interactive
Intelligence recommends readers to engage

the
service of an Interactive Intelligence MIC Certified
Engineer

or the manufacturers of the equip
ment(s) described
with
in to assist with

the planning and
deployment of
Messaging Interaction Center
.




2

Chapter 1:
General Information

Components

PBX or
IP
-
PBX

PBX Vendor

Mitel

Model

3300 ICP MX

Software Version

7

Telephony Signaling

T1 Q.SIG

Additional Notes

None

VoIP
Gateway

Gateway Vendor

AudioCodes

Model

Mediant 2000

Software Version

5.20A.037.0
01

VoIP Protocol

SIP

Additional Notes

Tests were conducted with Mediant 2000
. However
,

these tests are

also applicable to
Mediant 1000
.

Interactive Intelligence Messaging Interaction Center

Version

2.4 SU 25 + MIC SU 6

Prerequisites

Gateway

Prerequisit
es



The Mitel PBX must have the same source number for MWI ON (Lamp on) and MWI OFF (Lamp
off). Therefore, it's required to configure the gateway parameter
M
wiSource
N
umber

as the
voicemail pilot number (i.e., 2690 in our testing environment example). This n
umber is shown
on the subscribers phone display when the MWI lamp is turned on.

PBX Prerequisites



PBX with T1 dual framer module.



T1 QSIG option
.



3

Cabling Requirements



This integration uses
cross
-
over

RJ
-
48c cables

(pairs 1,

2 and 4,

5 crossed)
to connect d
igital
trunks (T1/E1) between and Mediant 2000 trunk interfaces
.

Summary and Limitations



A check in this box indicates the UM feature set is fully functional when using the PBX/gateway in
question.




4

Chapter 2:

Gateway Setup

Step 1:

SIP E
nvironment Setup



5

Step 2: Routing Setup




Note:

The Proxy IP Address must be one that corresponds to the network environment in which the
MIC server is installed (For example, 10.15.3.207 or the FQDN of the MIC host).



6

Step 3: Coder Setup






7

S
tep 4: Digit Collection Setup





8

Step 5:
Supplementary Services Setting






















Note:
Choose any 4
-
digit number that is not used in the PBX for Transfer Prefix (e.g., 9989).


9


Step
6
: T
runk Group Setup













10

Step
7
:
TDM Bus Setting





































11


Step
8
: Trunk Setting Setup



















12

Step
9
:
Fax Setup


























13

Step 1
0
:
Voicemail

Setup



























14

Step 1
1
: General Setup





15

Step 1
2
: General Setup (Cont.)

The following configuration items must be set in the INI file, or via the AdminPage web interface
(
http://gateway/AdminPage
). You will see these in t
he attached INI example.


ISDNIBehavior = 1073741824

EnableMWI = 1

SubscriptionMode = 1

EnableDetectRemoteMACChange = 2

ECNLPMode = 1

MwiSourceNumber = 2690


Note:

This parameter should be set to the Voice Mail pilot number (See Step 9 on PBX Setup).

Trun
kTransferMode_
X

=
0

Note:

"X"
refer
s to the Trunk number, for example: for the first trunk

TrunkTransferMode_0 =
0




16

Step 13
: Reset Mediant 2000


Click
Reset

to reset the gateway.

Configuration Files





17

Chapter 3: PBX Setup

Informatio
n used for this test case:



Digital VoiceMail ports:
T1 Q.SIG Trunk



VoiceMail Hunt Group Pilot: ext.
2690



VoiceMail User Phone: ext.
2608

and ext.
8999

Step 1:
Ensure PBX is equipped with T1 PRI Module to Support
T1 QSIG


Use the following path to ensure th
at the PBX is equipped with a T1 PRI module:

System Configuration

/
UnitsModules

/
Units Configuration Display.




Step 2:

Create Class of Service for QSIG Trunks


Use the following path to create CoS for QSIG trunk:

System Configuration

/
Trunks

/
Class
of Service Options Assignment
.



18





19

Change the following options to 'Yes':

ANI/DNIS/ISDN Number Delivery Trunk

Call Announce Line

Call Forwarding (External Destination)

Call Hold
-

Retrieve with Hold Key

Call Reroute after CFFM to busy Destination





20

Chang
e the following options to 'Yes':

Display ANI/DNIS/ISDN Calling/Called Number

Display DNIS/Called Number Before Digit Modification

Display Dialed Digits During Outgoing Calls

Display Held Call ID on Transfer





21

Change the following options to 'Yes':

Messa
ge Waiting
-

Audible Tone Notification

ONS CLASS/CLIP: Message Waiting Activate/Deactivate

ONS CLASS/CLIP Set

Public Network Access via DPNSS

Public Network Identity Provided

Public Network to Public Network Connection Allowed

Public Trunk

R2 Call Progress

Tone





22

Change the following options to 'Yes':

Third Party Call Forward Follow Me Accept

Third Party Call Forward Follow Me Allow

Trunk Flash Allowed







23

Step 3:
Create Link Descriptor Assignment


Use the following path to create Link Descriptor Assign
ment:

System Configuration

/
Trunks

/
Digital Trunks

/
ISDN
-
PRI

/
Link Descriptor Assignment
.






24

Step 4:

Create a Digital Link Assignment


Use the following path to

create a Digital Link Assignment:

System Configuration

/
Trunks

/
Digital Trunks

/
ISDN
-
P
RI

/

Digital Link Assignment
.






25



26

Step 5:

Add MSDN
-
DPNSS
-
DASSII Trunk Circuit Descriptor


Use the following path to

Add MSDN
-
DPNSS
-
DASSII Trunk Circuit Descriptor:

System Configuration

/
Trunks

/
Digital Trunks

/
ISDN
-
PRI/MSDN
-
DPNSS
-
DASSII Trunk
Circuit
Descriptor
.






27



28

Step 6:

Create a Trunk Service Assignment


Use the following path to

create a Trunk Service Assignment:

System Configuration

/
Trunks

/
Digital Trunks

/
ISDN
-
PRI

/
Trunk Service Assignment
.







29

Step 7:
Assign Digital Trunks


Use the fol
lowing path to

assign Digital Trunks:

System Configuration

/
Trunks

/
Digital Trunks

/
ISDN
-
PRI

/
Digital Trunk Assignment
.






30



31

Step 8:

Add Trunk Group for QSIG


Use the following path to

assign ARS for call routing:

System Administration

/
Automatic R
oute Selection

/
Trunk Group Assignment
.


Add a new trunk group for QSIG and add members to this group.




32

Step 9:

Add Route Assignment


Use the following path to

a
dd Route Assignment:

System Administration

/
Automatic Route Selection

/
Route Assignment
.






33



34

Step 10:
Add ARS Digits Dialed Assignment


Use the following path to

add ARS Digits Dialed Assignment:

System Administration

/
Automatic Route Selection

/
ARS Digits Dialed Assignment
.







35



36


Step 11:

Option Assignment


In System Option Assignment,
change the following fields:

Route Optimization Network ID: change to any unique number.

DPNSS/QSIG Diversion Enabled: change to 'Yes'.




Fail
-
Over Configuration

N/A.

Tested Phones

Mitel SuperSet 4025

Other Comments

None.



37

Chapter 4:

Messaging Interactio
n Center 2.4
Validation Test Matrix

Testing the Core Feature Set

The following table contains a set of tests for assessing the functionality of the UM core feature set.
The results are recorded as either:



Pass (
P
)



Conditional Pass (
CP
)



Fail (
F
)



Not Tested
(
NT
)



Not Applicable (
NA
)

Refer to:



Appendix for a more detailed description of how to perform each call scenario.



Section 6.1 for detailed descriptions of call scenario failures, if any.


No.

Call Scenarios (see appendix for
more detailed instructions)

(P/
CP/F/NT)

Reason for Failure (see 6.1 for more
detailed descriptions)

1

Dial the pilot number from a phone
extension that is NOT enabled for Unified
Messaging and logon to a user’s mailbox.


Confirm hearing the prompt: “Welcome
to Commun楴é⸠To access you
r ma楬ioxⰠ
enter your extension…”

P


2

Navigate mailbox using Mobile Office

NT


3

Navigate mailbox using the Telephony
User Interface (TUI).

P


4

Dial user extension and leave a
voicemail.


4a

Dial user extension and leave a voicemail
from an internal
extension.

P


4b

Dial user extension and leave a voicemail
from an external phone.

P


5

Dial Auto Attendant (AA).


Dial the extension for the AA and confirm
the AA answers the call.

P


6

Call Transfer by Dial By Name.



6a

Call Transfer by Dial By Name

and have
the called party answer.


Confirm the correct called party answers
the phone.

P




38

No.

Call Scenarios (see appendix for
more detailed instructions)

(P/
CP/F/NT)

Reason for Failure (see 6.1 for more
detailed descriptions)

6b

Call Transfer by Dial By Name when the
called party’s phone is busy.


Conf楲m the ca汬l楳 routed to the ca汬ld
party’s voicemail.

P


6c

Call Transfer by Dial

by Name when the
called party does not answer.


Confirm the call is routed to the called
party’s voicemail.

P


7

Configure a button on the phone of a UM
-
enabled user to forward the user to the
pilot number. Press the voicemail button.


Confirm you are

sent to the prompt:
“Welcometo Communité. Please enter
your passcode.”

P


8

Send a test
Fax

message to user
extension.


Confirm the
Fax

is received in the user’s
楮iox.

NT


9

Setup Message Waiting Indicator (MWI).


P



Detailed Description of Limitat
ions


Failure Point

None

Phone type (if phone
-
specific)


Call scenarios(s) associated with failure point


List of UM features affected by failure point


Additional Comments










39

Chapter 5:

Troubleshooting

The tools used for debugging include network

sniffer applications (such as Ethereal) and AudioCodes'
Syslog protocol.

The Syslog client, embedded in the AudioCodes gateways (MP
-
11x, Mediant 1000, and Mediant 2000),
sends error reports/events generated by the gateway application to a Syslog server, u
sing IP/UDP
protocol.

To activate the Syslog client on the AudioCodes gateways:

1.

Set the parameter
Enable Syslog

to 'Enable'.

2.

Use the parameter
Syslog Server IP Address

to define the IP address of the Syslog server you
use.


Note:

The Syslog Ser
ver IP address must be one that corresponds to your network environment in
which the Syslog server is insta
lled (for example, 10.15.2.5).


3.

To determine the Syslog logging level,
set

the
Debug Level

parameter to
5
.

4.

Change the
CDR Report Level

to 'End

Call' to enable additional call information.

Step 2

Step 1



40



AudioCodes has also developed
advanced diagnostic

tools that may be used for high
-
level
troubleshooting. These tools include the following:



PSTN Trace:

PSTN Trace is a procedure used to monitor and tra
ce the PSTN elements
(E1/T1) in AudioCodes digital gateways (Mediant 1000 & Mediant 2000). These utilities are
designed to convert PSTN trace binary files to textual form.



DSP Recording:

DSP recording is a procedure used to monitor the DSP operation (e.g.
, rtp
packets and events).

Step
4

Step
3

41

Appendix

Dial Pilot Number and Mailbox Login

1.

Dial the pilot number of the MIC server from an extension that is NOT enabled for
Voicemail.

2.

Confirm hearing the greeting prompt: “Welcome to Communité. Please enter your
exten
sion...”

3.

Enter the extension, followed by the pound sign, and then the passcod of a Voicemail
enabled user.

4.

Confirm successful logon to the user’s mailbox.

Navigate Mailbox using Mobile Office

1.

Logon to a user’s mailbox who is licensed for Mobile O
ffice

2.

Navigate through the mailbox and try out various voice commands to confirm that the
Mobile Office is working properly.

3.

This test confirms that the RTP is flowing in both directions and speech recognition is
working properly.

Navigate Mailbox us
ing Telephony User Interface (TUI)

1.

Logon to a user’s mailbox.

2.

Navigate through the mailbox and try out the various key commands to confirm that the
TUI is working properly.

3.

This test confirms that both the voice RTP and DTMF RTP (RFC 2833) are flo
wing in both
directions.

Dial User Extension and Leave Voicemail

Note:

If you are having difficulty reaching the user’s voicemail, verify that the coverage path
for the user’s phone is set to the pilot number of the MIC server.

From an Internal Extension

1
.

From an internal extension, dial the extension for a Voicemail enabled user and leave a
voicemail message.

2.

Confirm the voicemail message arrives in the called user’s inbox.

3.

Confirm this message displays a valid MIC user’s name as the sender of this

voicemail.

From an External Phone

1.

From an external phone, dial the extension for a Voicemail enabled user and leave a
voicemail message.

2.

Confirm the voicemail message arrives in the called user’s inbox.

3.

Confirm this message displays the phone num
ber as the sender of this voicemail.



42

Dial Auto Attendan
t (AA)

1.

Create an Auto Attendant using the MIC Web Administrator:

2.

Dial the extension of Auto Attendant.

3.

Confirm the AA answers the call.

Call Transfer by Dial By Name

1.

Dial the pilot number f
or the MIC server from a phone that is NOT associated with a MIC
user.

2.

To search for a user by name:



Press 2 to Dial By Name.



Call Transfer by Dial By Name by entering the name of an MIC user using the
telephone keypad, last name first.

Note:

Even thoug
h some keys are associated with three or four numbers, for each
letter, each key only needs to be pressed once regardless of the letter you want.
Ignore spaces and symbols when spelling the name.Called Party Answers



Call Transfer by Dial By Name to a user
in the same dial plan and have the called
party answer.

3.

Confirm the call is transferred successfully.

Called Party is Busy

1.

Call Transfer by Dial By Name to a user in the same dial plan when the called party is
busy.

2.

Confirm the calling user is rou
ted to the correct voicemail.

Called Party does not Answer

1.

Call Transfer by Dial By Name to a user in the same dial plan and have the called party not
answer the call.

2.

Confirm the calling user is routed to the correct voicemail.

Voicemail Button

1.

C
onfigure a button on the phone of a Voicemail enabled user to route the user to the pilot
number of the MIC server.

2.

Press the voicemail button.

3.

Confirm you are sent to the prompt: “Welcome to Communité. Please enter your
passcode...”

Note:

If you ar
e not hearing this prompt, verify that the button configured on the phone
passes the user’s extension as the redirect number. This means that the user extension
should appear in the diversion information of the SIP invite.

Testing
Fax

Features

To test fax
functionality:

1.

Dial the extension for a fax
-
enabled MIC user from a fax machine.

2.

Confirm the fax message is received in the user’s inbox.



43

Note:

You may notice that the MIC server answers the call as though it is a voice call (i.e.
you will hear: “Ple
ase leave a message for…”). When the MIC server detects the fax CNG
tones, it switches into fax receiving mode, and the voice prompts terminate.

Note:

MIC only support
s

T.38 for sending fax.

Message Waiting Indicator (MWI)

1.

Enable MWI for a Voicemail en
abled user.

2.

Leave a message for that user.

3.

Verify MWI goes on

4.

Delete or Mark Saved that message

5.

Verify MWI goes off

Note:

MWI doesn’t go off until there are no more New messages in the Inbox.



44

Appendix A:
Change Log


Change Log Date

Change
s Made

05
-
07
-
2008

Created document.

05
-
19
-
2008

Cleaned up formatting and added front matter.

06
-
05
-
2008

Completed Mitel information




























45

Appendix B: Acronyms Used in This Document

Here are some of the most important acronyms u
sed in this document.


CA
S

Centralized Attendant Service

DTMF

Dual Tone Multi
-
Frequency

IA

Interaction Administrator

IC

Interaction Center

IP

Internet Protocol

PBX

Private Branch Exchange

SIP

Session Initiation Protocol

TDM

Time Division Multiplexing

VoIP

Voice Over IP (Voice Over Internet Protocol)