Interactive Intelligence Messaging Interaction Center 2.4

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 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.

Copyright ©1994 – 2008 Interactive Intelligence Inc. All rights reserved. Interactive Intelligence®, Vonexus®, Interaction Center Platform®, Communité®, Vonexus Enterprise Interaction Center®, Interactive Intelligence 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, Interaction 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 Voice-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.

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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

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: TDM 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

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

Chapter 4: Messaging Interaction Center 2.4 Validation Test Matrix 37

Testing the Core Feature Set 37

Detailed Description 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: Change Log 44

Appendix B: Acronyms Used in This Document 45

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 / 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.037.001
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 Prerequisites

· 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 MwiSourceNumber as the voicemail pilot number (i.e., 2690 in our testing environment example). This number 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.

Cabling Requirements

· This integration uses cross-over RJ-48c cables (pairs 1, 2 and 4, 5 crossed) to connect digital 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.

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: Supplementary Services Setting

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

Step 6: Trunk Group Setup

Step 7: TDM Bus Setting

Step 8: Trunk Setting Setup

Step 9: Fax Setup

Step 10: Voicemail Setup

Step 11: General Setup

Step 12: 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.

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).

TrunkTransferMode_X = 0

Note: "X" refers to the Trunk number, for example: for the first trunk
TrunkTransferMode_0 = 0

Step 13: Reset Mediant 2000

Click Reset to reset the gateway.

Configuration Files

Chapter 3: PBX Setup

Information 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 that 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.

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

Change 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

Change the following options to 'Yes':

Message 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

Change the following options to 'Yes':

Third Party Call Forward Follow Me Accept

Third Party Call Forward Follow Me Allow

Trunk Flash Allowed

Step 3: Create Link Descriptor Assignment

Use the following path to create Link Descriptor Assignment:

System Configuration / Trunks / Digital Trunks / ISDN-PRI / Link Descriptor Assignment.

Step 4: Create a Digital Link Assignment

Use the following path to create a Digital Link Assignment:

System Configuration / Trunks / Digital Trunks / ISDN-PRI / Digital Link Assignment.

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.

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.

Step 7: Assign Digital Trunks

Use the following path to assign Digital Trunks:

System Configuration / Trunks / Digital Trunks / ISDN-PRI / Digital Trunk Assignment.

Step 8: Add Trunk Group for QSIG

Use the following path to assign ARS for call routing:

System Administration / Automatic Route Selection / Trunk Group Assignment.

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

Step 9: Add Route Assignment

Use the following path to add Route Assignment:

System Administration / Automatic Route Selection / Route Assignment.

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.

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.

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) / (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 Communité. To access your mailbox, 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
6b / Call Transfer by Dial By Name when the called party’s phone is busy.
Confirm the call is routed to the called 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 inbox. / NT
9 / Setup Message Waiting Indicator (MWI). / P

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

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.