Interactive Intelligence Messaging Interaction Center 2.

4
PBX Configuration Note:
Avaya S8300 with AudioCodes Mediant 2000 using T1 CAS (In-band DTMF Tones)

Technical Reference
By AudioCodes
READ THIS BEFORE YOU PROCEED
This document is for informational purposes only 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 configuration required to setup Avaya S8300 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.

Interactive Intelligence, Inc.

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Indianapolis, Indiana 46278
Telephone/Fax (317) 872-3000
www.ININ.com

Copyright and Trademark Information

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Interaction Administrator, Interaction Attendant, Interaction Client, Interaction Designer,
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Telephone/Fax (317) 872-3000
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Interaction Center Platform Statement

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

Customer Interaction Center (CIC)

Vonexus Enterprise Interaction Center (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.

Table of Contents
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
PBX Prerequisites.................................................................................................................................... 2
Cabling Requirements.............................................................................................................................. 2

Summary and Limitations........................................................................................................... 2

Chapter 2: Gateway Setup............................................................................................................ 3
Step 1: SIP Environment Setup.................................................................................................. 3
3
Step 2: Routing Setup ............................................................................................................... 4
Step 3: Coder Setup................................................................................................................... 5
Step 4: Digit Collection Setup .................................................................................................... 6
Step 5: Manipulation Routing Setup........................................................................................... 7
Step 6: Trunk Group Setup........................................................................................................ 8
Step 7: TDM Bus Settings.......................................................................................................... 9
Step 8: CAS Table Download................................................................................................... 10
Step 9: Trunk Setting Setup..................................................................................................... 11
Step 10: Voicemail In-Band DTMF Setup................................................................................12
Step 11: Fax Setup ................................................................................................................. 13
Step 12: General Setup............................................................................................................ 14
Step 13 : General Setup (Cont.)...............................................................................................15
Step 14: Reset Mediant 2000 .................................................................................................. 16
Configuration Files.................................................................................................................................. 16

Chapter 3: PBX Setup................................................................................................................. 17

Step 1: Add Station with Coverage Path..................................................................................17
Step 2: Define the Coverage Path Properties .........................................................................18
Step 3: T1 Setting using E1/T1 Card........................................................................................19
Step 4: Configuring T1 Ports.................................................................................................... 20
Step 5: Add a Hunt Group........................................................................................................ 21
Step 6: Assign T1 Ports to Hunt Group....................................................................................22
Step 7: Viewing Mode-Code Related System Parameters and Feature Access Codes...........23

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Feature Access Code.............................................................................................................................24

Fail-Over Configuration.......................................................................................................................... 24
Tested Phones....................................................................................................................................... 24
Other Comments.................................................................................................................................... 24

Chapter 4: Messaging Interaction Center 2.4 Validation Test Matrix.....................................25

Testing the Core Feature Set................................................................................................... 25
Detailed Description of Limitations...........................................................................................26
Chapter 5: Troubleshooting....................................................................................................... 27
Appendix...................................................................................................................................... 29
Dial Pilot Number and Mailbox Login.......................................................................................29
Navigate Mailbox using Mobile Office......................................................................................29
Navigate Mailbox using Telephony User Interface (TUI)..........................................................29
Dial User Extension and Leave Voicemail...............................................................................29
From an Internal Extension.................................................................................................................... 29
From an External Phone......................................................................................................................... 29

Dial Auto Attendant (AA).......................................................................................................... 30

Call Transfer by Dial By Name.................................................................................................30
Called Party is Busy............................................................................................................................... 30
Called Party does not Answer................................................................................................................ 30

Voicemail Button...................................................................................................................... 30
Testing Fax Features............................................................................................................... 30
Message Waiting Indicator (MWI)............................................................................................31
Appendix A: Change Log........................................................................................................... 32
Appendix B: Acronyms Used in This Document......................................................................33

iii

Who should read this document

This document is intended for Systems Integrators with significant telephony 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 configuration 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 equipment(s) described within to assist with the planning and
deployment of Messaging Interaction Center.

Chapter 1: General Information

Components
PBX or IP-PBX
PBX Vendor

Avaya

Model

S8300

Software Version

R013x.01.2.632.1

Telephony Signaling

T1 CAS In-band DTMF Tones

Additional Notes

None

VoIP Gateway
Gateway Vendor

AudioCodes

Model

Mediant 2000

Software Version

5.00.017.003

VoIP Protocol

SIP

Interactive Intelligence Messaging Interaction Center

Version

2.4 SU 25 + MIC SU 6

Prerequisites
PBX Prerequisites
Enter the command display system-parameters customer-options and then ensure that the Mode
Code for Centralized Voice Mail? parameter (on screen 4 of the PBX configuration tool) is set to y.
The PBX hardware must include an installed Trunk Card Module MM710.

Cabling Requirements
This integration uses standard RJ-48c cable to connect digital trunk (T1/E1) between MM710 and
Mediant 2000 Trunk interface.

Summary and Limitations

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

Chapter 2: Gateway Setup

Step 1: SIP Environment Setup

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).

Step 3: Coder Setup

Step 4: Digit Collection Setup

Step 5: Manipulation Routing Setup

Note: The Avaya In Band DTMF integration sends a DTMF sequence when dialing the pilot number to
login to the Voicemail TUI. This sequence must be manipulated to send the appropriate VM Pilot
Number to the MIC system. In the example above, the pilot number is 5999. The digits that are sent
by the Avaya are #02#callingparty##.

Step 6: Trunk Group Setup

Note: The phone number here is for debugging purposes only and can be any number.

Step 7: TDM Bus Settings

Step 8: CAS Table Download

Note: The CAS file must be downloaded to the Mediant. A suitable CAS file is attached to this
document (refer to Section 4.1).

10

Step 9: Trunk Setting Setup

11

Step 10: Voicemail In-Band DTMF Setup

Note: These sequences may vary from one site to site. You will need ot obtain the correct DTMF
sequences from the Avaya PBX Administrator at the install location.

12

Step 11: Fax Setup

13

Step 12: General Setup

14

Step 13 : 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 the attached INI example.

EnableMWI = 1
SubscriptionMode = 1
LineTransferMode = 1
IsSpecialDigits = 1
EnableDetectRemoteMACChange = 2
ECNLPMode = 1
TrunkTransferMode_X = 3
Note: "X" refers to the Trunk number, for example: for the first trunk
TrunkTransferMode_0 = 3

15

Step 14: Reset Mediant 2000

Click Reset to reset the gateway.

Configuration Files
The ZIP file includes the following files:
Audiocodes configuration ini file (.ini file extension)
Audiocodes T1 CAS file (.dat file extension)
M2K Avaya s8300 T1 CAS DTMF MIC Sample.zip

MICAvayaSampleFile
s.zip

16

Chapter 3: PBX Setup

Information used for this test case:
Digital VoiceMail ports: ext. 5061 5084 ,24 T1 Channel
VoiceMail Hunt Group Pilot: ext. 5060
VoiceMail User Phone: ext. 5006 and ext. 5096
Coverage Path for ext. 5006 and 5096 is 2
User Test Phone Avaya 2420

Step 1: Add Station with Coverage Path

Add a station using the Add Station command, and then configure the station using the below
settings (in this example, it's a digital station, but it can also be an analog station). Define the
Coverage Path for this station, for example, as 2.

17

Step 2: Define the Coverage Path Properties

Add a coverage path using the Add Coverage Path command. Configure the coverage path
properties using the below settings. Note: Ensure that you define the coverage destination point.

The coverage path example configured in the figure above, means that when the station does not
answer the call or the call is busy, the call is forwarded to hunt group 2 (Point1: h2).

18

Step 3: T1 Setting using E1/T1 Card

Define T1 using the E1/T1 card that is connected to the Mediant 2000, using the Add DS1 command.

19

Step 4: Configuring T1 Ports

Configure the ports on the T1 circuit. Note that the parameter Type must be set to VMIFD to enable
DTMFs. For T1 define 24 stations.

20

Step 5: Add a Hunt Group

Add a hunt group using the Add Hunt Group command. Define the hunt group number (for example,
3) and properties using the below settings.

21

Step 6: Assign T1 Ports to Hunt Group

Assign the T1 ports that you defined in Step 4 to the hunt group.

22

Step 7: Viewing Mode-Code Related System Parameters and

Feature Access Codes
The DTMF signals sent from the PBX to the voice mail system providing the forwarding information
(calling party, called party, etc.,) are called Mode Code.
The figure below displays the mode-code related system parameters for the Avaya S8300 PBX.

23

Feature Access Code

Below is the feature access code that defines the code sent from the gateway to the PBX for activating
/ deactivating Message Waiting Indication (MWI), for example, *16 and #16.

Fail-Over Configuration
N/A.

Tested Phones
Avaya 2420 Model Telephone.

Other Comments
None.

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Chapter 4:

Messaging Interaction 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)

Dial the pilot number from a phone

extension that is NOT enabled for Unified
Messaging and logon to a users mailbox.

(P/CP/F/NT)

Reason for Failure (see 6.1 for more

detailed descriptions)

Confirm hearing the prompt: Welcome

to Communit. To access your mailbox,
enter your extension
2

Navigate mailbox using Mobile Office

NT

Navigate mailbox using the Telephony

User Interface (TUI).

Dial user extension and leave a

voicemail.

4a

Dial user extension and leave a voicemail

from an internal extension.

4b

Dial user extension and leave a voicemail

from an external phone.

Dial Auto Attendant (AA).

Dial the extension for the AA and confirm

the AA answers the call.
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

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No.

Call Scenarios (see appendix for

more detailed instructions)

(P/CP/F/NT)

Reason for Failure (see 6.1 for more

detailed descriptions)

the phone.
6b

Call Transfer by Dial By Name when the

called partys phone is busy.

Confirm the call is routed to the called

partys voicemail.
6c

Call Transfer by Dial by Name when the

called party does not answer.

Confirm the call is routed to the called

partys voicemail.
7

Configure a button on the phone of a UMenabled user to forward the user to the
pilot number. Press the voicemail button.

Confirm you are sent to the prompt:

Welcometo Communit. Please enter
your passcode.
8

Send a test Fax message to user

extension.

NT

Confirm the Fax is received in the users

inbox.
9

Setup Message Waiting Indicator (MWI).

Detailed Description of Limitations

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

26

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, using 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.

Step 2
Step 1

Note: The Syslog Server IP address must be one that corresponds to your network environment in
which the Syslog server is installed (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.

27

Step 4
Step 3

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 trace 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).

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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 Communit. Please enter your
extension...

3.

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

4.

Confirm successful logon to the users mailbox.

Navigate Mailbox using Mobile Office

1.

Logon to a users mailbox who is licensed for Mobile Office

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 using Telephony User Interface (TUI)

1.

Logon to a users 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 flowing in both
directions.

Dial User Extension and Leave Voicemail

Note: If you are having difficulty reaching the users voicemail, verify that the coverage path
for the users 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 users inbox.

3.

Confirm this message displays a valid MIC users 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 users inbox.

3.

Confirm this message displays the phone number as the sender of this voicemail.

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Dial Auto Attendant (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 for 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 though 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 routed 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.

Configure 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 Communit. Please enter your
passcode...
Note: If you are not hearing this prompt, verify that the button configured on the phone
passes the users 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 users inbox.

30

Note: You may notice that the MIC server answers the call as though it is a voice call (i.e.
you will hear: Please 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 supports T.38 for sending fax.

Message Waiting Indicator (MWI)

1.

Enable MWI for a Voicemail enabled 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 doesnt go off until there are no more New messages in the Inbox.

31

Appendix A: Change Log

Change Log Date

Changes Made

05-07-2008

Created document.

05-19-2008

Cleaned up formatting and added front matter.

32

Appendix B: Acronyms Used in This Document

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

CAS

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)

33