Interactive Intelligence Messaging Interaction Center 2.4PBX Configuration Note:

Siemens HiPath 4000 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.

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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.

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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 Contents

Who should read this document.................................................................................................1

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

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

Components................................................................................................................................ 2PBX or IP-PBX........................................................................................................................2

VoIP Gateway........................................................................................................................2

Interactive Intelligence Messaging Interaction Center..........................................................2

Prerequisites............................................................................................................................... 2PBX Prerequisites..................................................................................................................2

Cabling Requirements..........................................................................................................2

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: Message Waiting Indication 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: General Setup.............................................................................................................13

Step 11: General Setup (Cont.).................................................................................................14

Step 12: Reset Mediant 2000....................................................................................................15Configuration Files..............................................................................................................15

Chapter 3: PBX Setup.................................................................................................................16Fail-Over Configuration.......................................................................................................24

Tested Phones....................................................................................................................25

Other Comments................................................................................................................25

Chapter 4: Messaging Interaction Center 2.4 Validation Test Matrix...............................26

Testing the Core Feature Set....................................................................................................26

Detailed Description of Limitations............................................................................................27

Chapter 5: Troubleshooting.................................................................................................28

Appendix...................................................................................................................................... 30

Dial Pilot Number and Mailbox Login........................................................................................30

Navigate Mailbox using Mobile Office.......................................................................................30

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Navigate Mailbox using Telephony User Interface (TUI)...........................................................30

Dial User Extension and Leave Voicemail................................................................................30From an Internal Extension.................................................................................................30

From an External Phone.....................................................................................................30

Dial Auto Attendant (AA)...........................................................................................................31

Call Transfer by Dial By Name..................................................................................................31Called Party is Busy............................................................................................................31

Called Party does not Answer.............................................................................................31

Voicemail Button....................................................................................................................... 31

Testing Fax Features................................................................................................................31

Message Waiting Indicator (MWI).............................................................................................32

Appendix A: Change Log...........................................................................................................33

Appendix B: Acronyms Used in This Document......................................................................34

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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.

1

Chapter 1: General Information

Components

PBX or IP-PBX

PBX Vendor Siemens

Model HiPath 4000

Software Version Ver 3.0 SMR5 SMP4

Telephony Signaling T1 Q.SIG

Additional Notes None

VoIP Gateway

Gateway Vendor AudioCodes

Model Mediant 2000

Software Version 5.0A.023.001

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

PBX Prerequisites Verify that PSS1V2 QSIG protocol is available by typing this command: dis-

prode:db,vartab.

The PBX is installed with Trunk card module Q2216-X DIU2U-M T1.

Cabling Requirements

This integration uses a standard RJ-48c cable to connect a digital trunk (T1/E1) between the Q2216-X DIU2U-M T1 card and Mediant 2000 trunk interface.

2

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

question.

3

Chapter 2: Gateway Setup

Step 1: SIP Environment Setup

4

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).

5

Step 3: Coder Setup

6

Step 4: Message Waiting Indication Setup

7

Step 5: Supplementary Services Setting

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

8

Step 6: Trunk Group Setup

9

Step 7: TDM Bus Setting

Step 8: Trunk Setting Setup

10

Step 9: Fax Setup

11

12

Step 10: General Setup

13

Step 11: 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.

VoiceMailInterface = 3 (for QSIG)

ISDNIBehavior = 1073741824

EnableMWI = 1

SubscriptionMode = 1

EnableDetectRemoteMACChange = 2

ECNLPMode = 1

TrunkTransferMode_X = 0

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

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Step 12: Reset Mediant 2000

Click Reset to reset the gateway.

Configuration Files

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Chapter 3: PBX SetupInformation used for this test case:

Our test configuration uses extensions 5000 to 5050 that are configured as off-premises extensions and are reachable through Trunk Group 3 configured for T1 QSIG. Calls to Siemens Optiset phone ext. 2417 are forwarded under busy/no answer conditions to ext. 5000.

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Step 1: T1 QSIG Trunk Group Setup

<disp-buendTGRP = 3FORMAT = l;DISP-BUEND:3,L;H500: AMO BUEND STARTED

+------------------------------- FORMAT = L -----------------------------------+| TGRP NUMBER : 3 TGRP NAME : T1 QSIG MAXIMUM NO. : 23 || CHARCON : NEUTRAL || SUBGROUP NO.: 4 DEVICE TYPE : S1CONN TRACENO : 0 || SEARCH MODE : CIRCULAR ACD THRESHOLD : * || NUMBER OF ASSOCIATED ROUTES : 1 PRIORITY : 2 || TDDRFLAG : ON TDDRTHRESHOLD: 2 SOURCEGROUPIDX : 1 || GDTRRULE : 0 ACDPMGRP : 0 || THE FOLLOWING TRUNKS (LTG-LTU-SLOT-CCT) HAVE BEEN ALLOCATED: |+------------------------------------------------------------------------------+| 1- 1- 14-0 1 | 1- 1- 14-0 2 | 1- 1- 14-0 3 || 1- 1- 14-0 4 | 1- 1- 14-0 5 | 1- 1- 14-0 6 || 1- 1- 14-0 7 | 1- 1- 14-0 8 | 1- 1- 14-0 9 || 1- 1- 14-0 10 | 1- 1- 14-0 11 | 1- 1- 14-0 12 || 1- 1- 14-0 13 | 1- 1- 14-0 14 | 1- 1- 14-0 15 || 1- 1- 14-0 16 | 1- 1- 14-0 17 | 1- 1- 14-0 18 || 1- 1- 14-0 19 | 1- 1- 14-0 20 | 1- 1- 14-0 21 || 1- 1- 14-0 22 | 1- 1- 14-0 23 | |+------------------------------------------------------------------------------+

AMO-BUEND-111 TRUNK GROUP DISPLAY COMPLETED;

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Step 2: Trunks SetupVerify that the trunks are configured with DEVICE TYPE: S1CONN and using the PSS1V2 protocol.

<disp-tdcsuPEN1 = 1-1-14-0;DISP-TDCSU:1-1-14-0;H500: AMO TDCSU STARTED +------------------------ DIGITAL TRUNK (FORMAT=L) -------------------------+ | DEV = S1CONN PEN = 1-01-014-0 TGRP = 3 | |---------------------------------------------------------------------------| | PROTVAR = PSS1V2 INS = Y SRCHMODE = CIR | | COTNO = 3 COPNO = 3 DPLN = 0 | | ITR = 0 COS = 3 LCOSV = 25 | | LCOSD = 1 CCT = DESTNO = 0 | | SEGMENT = 1 DEDSCC = DEDSVC = NONE | | FACILITY = DITIDX = SRTIDX = | | TRTBL = GDTR SIDANI = N ATNTYP = TIE | | CBMATTR = NONE NWMUXTIM = 10 TCHARG = N | | SUPPRESS = 0 DGTPR = CHIMAP = N | | ISDNIP = ISDNNP = | | PNPL2P = PNPL1P = PNPAC = | | TRACOUNT = 31 SATCOUNT = MANY NNO = 3 | | ALARMNO = 0 FIDX = 1 CARRIER = 1 | | ZONE = EMPTY COTX = 3 FWDX = 1 | | DOMTYPE = UNKNOWN DOMAINNO = 0 TPROFNO = | | INIGHT = CCHDL = | | UUSCCX = 16 UUSCCY = 8 FNIDX = 1 | | CLASSMRK = EC & G711 & G729OPT SRCGRP = | | TCCID = | |---------------------------------------------------------------------------| | BCNEG = N BCGR = 1 LWPAR = 1 | | LWPP = LWLT = LWPS = | | LWR1 = LWR2 = | | DMCALLWD = DMCSEC = | | SVCDOM = | | BCHAN = 1 && 23 | | | +---------------------------------------------------------------------------+ AMOUNT OF B-CHANNELS IN THIS DISPLAY-OUTPUT: 23

AMO-TDCSU-111 DIGITAL TRUNKS DISPLAY COMPLETED;

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Step 3: Trunk’s COP & COT Parameters

<disp-copCOPNO = 3;DISP-COP:3;H500: AMO COP STARTED

COP: 3 INFO: DEVICE: INDEP SOURCE: DB PARAMETER: LINE WITH START-DIAL-SIGNAL SDL SPECIAL MODE SFRM REGISTRATION OF LAYER 3 ADVISORIES L3AR START-DIALING-SIGNAL TIMER 1 (AMO DTIM2: PARAMETER STADIAL1) TIM1

AMO-COP -111 CLASS OF PARAMETER FOR DEVICE HANDLER DISPLAY COMPLETED; <disp-cotCOTNO = 3;DISP-COT:3;H500: AMO COT STARTED

COT: 3 INFO: DEVICE: INDEP SOURCE: DB PARAMETER: RECALL IF USER HANGS UP IN CONSULTATION CALL RCL UNRESTRICTED SUFFIX DIALING USD KNOCKING OVERRIDE POSSIBLE KNOR CALL EXTEND FOR BUSY, RING OR CALL STATE CEBC NETWORKWIDE AUTOMATIC CALLBACK ON BUSY CBBN NETWORKWIDE AUTOMATIC CALLBACK ON FREE CBFN NETWORKWIDE CALL FORWARDING PERMITTED FWDN NETWORKWIDE FORWARDING NO-ANSWER FNAN REGISTRATION OF IMPLAUSIBLE EVENTS IEVT END-OF-DIAL FOR BLOCK IS SET BLOC SEND NO NODE NUMBER TO PARTNER LWNC CONNECTION TO ROUTE OPTIMIZATION NODE ROPT INCOMING CIRCUIT FROM SYSTEM WITHOUT LCR NLCR TSC-SIGNALING FOR NETWORKWIDE FEATURES (MANDATORY) TSCS USE DEFAULT NODE NUMBER OF LINE DFNN INCOMING CIRCUIT FROM SYSTEM WITHOUT LCR (DATA) NLRD LINE WITH IMPLICIT NUMBERS LINO LAST REDIRECTING NUMBER IS SEND TO PHONEMAIL LRPM NO SIMPLE DIALOG AVAILABLE NOSD NO TONE NTON

AMO-COT -111 CLASS OF TRUNK FOR CALL PROCESSING DISPLAY COMPLETED;

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Step 4: Trunk Route's Information

<disp-richtMODE = ?MODE : ADDRESS-MODE OF THE ROUTECHARACTERISTIC : REQUIRED CONDITIONAL POSS. VALUES : ALL ALL PM PHONE MAIL INFO SERVICE INFORMATIONS LRTE LCR ROUTE NUMBER CD CODEMODE = all;DISP-RICHT:ALL;H500: AMO RICHT STARTED+------------------------------------------------------------------------------+| ROUTES FOR ALL DPLN SVC = VCE || |+-------+------------------+----+-+------------------------------+-----+----+--+|CODE |NAME, CQMAX, |TGRP|P| DTMF | LRTE|CPAR| F|| |DESTNO AND CPS |CCNO|L+---+---+----------------+-----+ | | W|| | 1 111112| |B|CNV|DSP| TEXT |PULS | | | D|| |12345 67890 123452| | | | | |PAUSE| | | B|+-------+------------------+----+-+---+---+----------------------+-----+----+--+|30 |..... ..... ......| 3| | W | W | | | 3| | ||NEUTRAL|VM | | | | | | | | | || |DNNO: 30 | | | | | | | | | || |PDNNO: 30 | | | | | | | | | || |DESTNO :30 | | | | | | | | | || |REROUT :YES | | | | | | | | | |

+-------+------------------+----+-+---+---+----------------+-----+-----+----+--+

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Step 5: Out-Dial Rule (ODR) Routing Setup

<disp-lodr;DISP-LODR;H500: AMO LODR STARTED +-------------------------------------------------------+ | ODR POSITION CMD PARAMETER | +--------+----------------------------------------------+ | 3 | 1 ECHO 3 | | | 2 ECHOALL | | | 3 END |

+--------+----------------------------------------------+

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Step 6: Route for Trunk Group 3 Setup

<disp-ldat;DISP-LDAT;H500: AMO LDAT STARTED

| LROUTE = 3 NAME = VM SERVICE = VCE || TYPE = NWLCR DNNO OF ROUTE = 30 || SERVICE INFO = |+------+-----+-----+----+-----+----------+-----------+----------+------+-------+| | | | | | SCHEDULE | CARRIER | | | ||LRTEL | LVAL|TGRP | ODR|LAUTH| ABCDEFGH | ZONE| LATTR | LDSRT|COTIDX |+------+-----+-----+----+-----+----------+-----------+----------+------+-------+| 1| 1| 3| 1| 1 | ******** | 1 EMPTY| NONE | | 0 |

+------------------------------------------------------------------------------+

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Step 7: Dial Plan Setup

<disp-wabeTYPE = gen;DISP-WABE:GEN;H500: AMO WABE STARTED ------------------------------------------------------------------------------ | DIGIT INTERPRETATION VALID FOR ALL DIAL PLANS | ------------------------------------------------------------------------------ | | CALL PROGRESS STATE | DIGIT | RESERVED/CONVERT | | CODE | 1 11111 11112 22| ANALYSIS | DNI/ADD-INFO | | |0 12345 67890 12345 67890 12| RESULT | *=OWN NODE | ------------------------------------------------------------------------------ | 30 |* ..... ..... ..... ..... ..| NETRTE | | | 5000 - 5050 |. .**** ***** **... ..... .*| STN | | | | | |DESTNO 30 | | | | |DNNO 0- 0- 30 | | | | |PDNNO 0- 0- 30 |

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Step 8: MWI Setup

<disp-richtMODE = pm;DISP-RICHT:PM;H500: AMO RICHT STARTED+------+----------------------+------------------------------+--------+ | IDX | SAN | NAME | TYPE | +------+----------------------+------------------------------+--------+| 1|5050 | |OTHER |+------+----------------------+------------------------------+--------+ <disp-sbcsuSTNO = 2417;DISP-SBCSU:2417;H500: AMO SBCSU STARTED

---------------------------- USER DATA --------------------------------------- STNO =2417 OPT =OPTI COS1 =5 DPLN =0 MAINO =2417 CONN =IP2 COS2 =5 ITR =0 PEN = 1- 1- 1- 9 LCOSV1 =25 COSX =0 INS =Y ASYNCT =500 LCOSV2 =25 PERMACT = LCOSD1 =1 SSTNO =N EXTBUS = LCOSD2 =1 CBKBMAX =5 TRACE =N RCBKB =N ALARMNO =0 DFSVCANA= SPDI =10 RCBKNA =N HMUSIC =0 FLASH = SPDC1 = CBKNAMB =Y PMIDX =1 SPDC2 = COMGRP =0 SECR =N DIGNODIS=N DSSTNA =N STD =52 CALLOG =NONE DSSTNB =Y TEXTSEL =(GERMAN)

REP =0 OPTICOM =N OPTIUSB : VPI = IDCR =N OPTICA = OPTIS0A :0 VCI = APPM = OPTIDA = OPTISPA :0 PATTERN = OPTIABA :0

DCFWBUSY=N HEADSET =N APMOBUSR=N APICLASS= DNIDSP =N HSKEY =NORMAL IPCODEC =G711PREF SECAPPL =0 DTMFBLK =N IPPASSW ="*****" DTMFCTRD=N BASICSVC= DVCFIG =OPTIIP * TSI =1 SPROT = SOPTIDX = DPROT = DOPTIDX = FPROT = FOPTIDX = ------------------- MOBILE HFA STATE ----------------------------------------- IP PHONE INACTIVE :N ------------------- ACTIVATION IDENTIFIERS FOR FEATURES ---------------------- HTOS :N DND :N HTOD :N VCP :N TWLOGIN :N HTOF :N CWT :N ------------------- FEATURES AND GROUP MEMBERSHIPS --------------------------- PUGR : ESSTN : KEYSYS :Y NOPTNO : SRCGRP :(1 ) TCLASS : 0 HUNT CD :N ------------------- SUBSCRIBER ATTRIBUTES (AMO SDAT) ------------------------- KN VC SPKRCALL COMSPK BROADCST ETD ------------------------------------------------------------------------------ *) OPTIPOINT420

AMO-SBCSU-111 STATION AND S0-BUS CONFIGURATION OF SWITCHING UNIT DISPLAY COMPLETED;

Fail-Over Configuration

N/A.

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Tested Phones

Analog Phones

Siemens Optiset phone.

Other Comments

None.

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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

27

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).

29

Step 4

Step 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 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 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 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 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 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 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.

31

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

32

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

33

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)

34