Interactive Intelligence Messaging Interaction Center 2.4PBX Configuration Note:

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

Technical Reference

By AudioCodes

READ THIS BEFORE YOU PROCEEDThis 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

ContentThis 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. 7601 Interactive Way

Indianapolis, Indiana 46278Telephone/Fax (317) 872-3000

www.ININ.com

Copyright and Trademark InformationCopyright ©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.veryPDF is Copyright © 2000-2005 by veryPDF, Inc. Other brand and/or product names referenced in this document are the trademarks or registered trademarks of their respective companies.

NOTICEThis 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 DAMAGES, INCLUDING ANY LOST PROFITS OR OTHER INCIDENTAL OR CONSEQUENTIAL DAMAGES RELATING TO THE PRODUCT, AND, (4) THIS AGREEMENT SHALL BE ENFORCEABLE BY TELCORDIA.

DISCLAIMERINTERACTIVE INTELLIGENCE (INTERACTIVE) HAS NO RESPONSIBILITY UNDER WARRANTY, INDEMNIFICATION OR OTHERWISE, FOR MODIFICATION OR CUSTOMIZATION OF ANY INTERACTIVE SOFTWARE BY INTERACTIVE, CUSTOMER OR ANY THIRD PARTY EVEN IF SUCH CUSTOMIZATION AND/OR MODIFICATION IS DONE USING INTERACTIVE TOOLS, TRAINING OR METHODS DOCUMENTED BY INTERACTIVE. Interactive Intelligence Inc. 7601 Interactive WayIndianapolis, Indiana 46278Telephone/Fax (317) 872-3000www.ININ.com

Interaction Center Platform StatementThis 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.

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Table of ContentsWho should read this document.................................................................................................1

Technical Support......................................................................................................................... 1

Chapter 1: General Information..................................................................................................2Components................................................................................................................................ 2

PBX or IP-PBX........................................................................................................................2VoIP Gateway........................................................................................................................2Interactive Intelligence Messaging Interaction Center..........................................................2

Prerequisites............................................................................................................................... 2Gateway Prerequisites..........................................................................................................2PBX Prerequisites..................................................................................................................2Cabling Requirements..........................................................................................................3

Summary and Limitations............................................................................................................3

Chapter 2: Gateway Setup.....................................................................................................4Step 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....................................................................................................16Configuration Files..............................................................................................................16

................................................................................................................16

Chapter 3: PBX Setup.................................................................................................................17Step 1: Ensure PBX is equipped with T1 PRI Module to Support T1 QSIG..............................17

Fail-Over Configuration.......................................................................................................36Tested Phones....................................................................................................................36Other Comments................................................................................................................36

Chapter 4: Messaging Interaction Center 2.4 Validation Test Matrix...............................37

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Testing the Core Feature Set....................................................................................................37

Detailed Description of Limitations............................................................................................38

Chapter 5: Troubleshooting.................................................................................................39

Appendix...................................................................................................................................... 41Dial 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................................................................................41From an Internal Extension.................................................................................................41From an External Phone.....................................................................................................41

Dial Auto Attendant (AA)...........................................................................................................42

Call Transfer by Dial By Name..................................................................................................42Called Party is Busy............................................................................................................42Called 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

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Who should read this documentThis document is intended for Systems Integrators with significant telephony knowledge.

Technical SupportThe 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.

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Chapter 1: General InformationComponents

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.

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Summary and Limitations A check in this box indicates the UM feature set is fully functional when using the PBX/gateway in

question.

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Chapter 2: Gateway SetupStep 1: SIP Environment Setup

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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).

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Step 3: Coder Setup

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Step 4: Digit Collection Setup

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Step 5: Supplementary Services Setting

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

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Step 6: Trunk Group Setup

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Step 7: TDM Bus Setting

Step 8: Trunk Setting Setup

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Step 9: Fax Setup

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Step 10: Voicemail Setup

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Step 11: General Setup

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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 = 1073741824EnableMWI = 1SubscriptionMode = 1EnableDetectRemoteMACChange = 2ECNLPMode = 1MwiSourceNumber = 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

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Step 13: Reset Mediant 2000

Click Reset to reset the gateway.

Configuration Files

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Chapter 3: PBX SetupInformation 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 QSIGUse 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 TrunksUse the following path to create CoS for QSIG trunk:System Configuration / Trunks / Class of Service Options Assignment.

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Change the following options to 'Yes':ANI/DNIS/ISDN Number Delivery TrunkCall Announce LineCall Forwarding (External Destination)Call Hold - Retrieve with Hold KeyCall Reroute after CFFM to busy Destination

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Change the following options to 'Yes':Display ANI/DNIS/ISDN Calling/Called NumberDisplay DNIS/Called Number Before Digit ModificationDisplay Dialed Digits During Outgoing CallsDisplay Held Call ID on Transfer

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Change the following options to 'Yes':Message Waiting - Audible Tone NotificationONS CLASS/CLIP: Message Waiting Activate/DeactivateONS CLASS/CLIP SetPublic Network Access via DPNSSPublic Network Identity ProvidedPublic Network to Public Network Connection AllowedPublic TrunkR2 Call Progress Tone

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Change the following options to 'Yes':Third Party Call Forward Follow Me AcceptThird Party Call Forward Follow Me AllowTrunk Flash Allowed

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Step 3: Create Link Descriptor AssignmentUse the following path to create Link Descriptor Assignment:System Configuration / Trunks / Digital Trunks / ISDN-PRI / Link Descriptor Assignment.

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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.

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Step 5: Add MSDN-DPNSS-DASSII Trunk Circuit DescriptorUse the following path to Add MSDN-DPNSS-DASSII Trunk Circuit Descriptor:System Configuration / Trunks / Digital Trunks / ISDN-PRI/MSDN-DPNSS-DASSII Trunk Circuit Descriptor.

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Step 6: Create a Trunk Service AssignmentUse the following path to create a Trunk Service Assignment:System Configuration / Trunks / Digital Trunks / ISDN-PRI / Trunk Service Assignment.

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Step 7: Assign Digital TrunksUse the following path to assign Digital Trunks: System Configuration / Trunks / Digital Trunks / ISDN-PRI / Digital Trunk Assignment.

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Step 8: Add Trunk Group for QSIGUse 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.

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Step 9: Add Route Assignment

Use the following path to add Route Assignment:System Administration / Automatic Route Selection / Route Assignment.

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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.

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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 ConfigurationN/A.

Tested PhonesMitel SuperSet 4025

Other CommentsNone.

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Chapter 4: Messaging Interaction Center 2.4 Validation Test Matrix

Testing the Core Feature SetThe 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

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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)

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

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Chapter 5: TroubleshootingThe 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.

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.

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Step 2

Step 1

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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Step 4Step 3

Appendix

Dial Pilot Number and Mailbox Login1. 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

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 user’s mailbox.

Navigate Mailbox using Mobile Office1. Logon to a user’s mailbox who is licensed for Mobile Office2. 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 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 flowing in both

directions.

Dial User Extension and Leave VoicemailNote: 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 Extension1. 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 Phone1. 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 number as the sender of this voicemail.

Dial Auto Attendant (AA)1. Create an Auto Attendant using the MIC Web Administrator:

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2. Dial the extension of Auto Attendant.3. Confirm the AA answers the call.

Call Transfer by Dial By Name1. 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 Busy1. 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 Answer1. 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 Button1. 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 Communité. Please enter your

passcode...”Note: If you are 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 FeaturesTo 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.

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.

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Message Waiting Indicator (MWI)1. Enable MWI for a Voicemail enabled user.2. Leave a message for that user.3. Verify MWI goes on4. Delete or Mark Saved that message5. Verify MWI goes off

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

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Appendix A: Change Log

Change Log Date Changes Made

05-07-2008 Created document.

05-19-2008 Cleaned up formatting and added front matter.

06-05-2008 Completed Mitel information

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Appendix B: Acronyms Used in This DocumentHere 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)

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