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Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Avaya IP Office with NET VX1200 or VX1800 using T1 QSIG

By Updated

: NET : 2/12/2009

READ THIS BEFORE YOU PROCEED
This document is for informational purposes only and is provided “AS IS”. Microsoft, its partners and vendors cannot verify the accuracy of this information and take no responsibility for the content of this document. MICROSOFT, ITS PARTNERS AND VENDORS MAKE NO WARRANTIES, EXPRESS, IMPLIED OR STATUTORY, AS TO THE INFORMATION IN THIS DOCUMENT.

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1. Document Overview Content
This document describes the configuration required to integrate Microsoft Exchange 2007 UM with Avaya models IP OFFICE using NET VX1200 or VX1800 Intelligent Voice Gateway with T1 QSIG as the telephony signaling protocol 1. It also contains the results of the interoperability testing of Microsoft Exchange Server 2007 Unified Messaging (UM) based on this setup.

Intended Audience
This document is intended for Systems Integrators with significant telephony knowledge.

Technical Support
The information contained within this document has been provided by Microsoft 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. Microsoft is unable to provide support or assistance with the configuration or troubleshooting of components described within. Microsoft recommends readers to engage the service of an Microsoft Exchange 2007 Unified Messaging Specialist or the manufacturers of the equipment(s) described within to assist with the planning and deployment of Exchange Unified Messaging.

Microsoft Exchange 2007 Unified Messaging (UM) Specialists
These are Systems Integrators who have attended technical training on Exchange 2007 Unified Messaging conducted by Microsoft Exchange Engineering Team. For contact information, visit here.

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Some possible protocols: 1. 2. 3. 4. 5. 6. 7. 8. 9. Analog lines with inband DTMF Analog lines with SMDI Digital Set Emulation (specify) Direct SIP connection T1 CAS with inband DTMF T1 CAS with SMDI E1 CAS with inband DTMF E1 CAS with SMDI T1 Q.SIG

10. E1 Q.SIG

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Version Information
Date of Modification 2/12/2009 Details of Modification Initial version of the document

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2. Components Information
2.1. PBX or IP-PBX
PBX Vendor Model Software Version Telephony Signaling Additional Notes Avaya IP OFFICE 403/406/412 series PBXs Small Office Edition 3.2.54 T1 QSIG

2.2. VoIP Gateway
Gateway Vendor Model Software Version VoIP Protocol NET VX1200 or VX1800 4.7 SIP

2.3. Microsoft Exchange Server 2007 Unified Messaging
Version SP1

3. Prerequisites
3.1. Gateway Requirements
? ? The gateway support SRTP and TLS for fully encrypted VOIP The gateway supports self signed and 3rd party signed certfifcates

3.2. PBX Requirements
? T1 card

3.3. Cabling Requirements
? The PBX is connected to the gateway using standard RJ-48c straight through T1 cables.

4. 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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5. Gateway Setup Notes
Please reference VX documentation for the initial Command Line configuration items that are common for all VX Gateway nodes at: http://www.net.com Step 1: Start VX Configuration Wizard by clicking on the “Run Wizard” button:

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Step 2: Select the Primary role for the node as “Basic Unified Messaging ISDN Gateway” by enabling the Radio button for this role and then click ‘Next”:

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Step 3: Enter at least the Preferred DNS Server and Alternate DNS Server if available and if using DNS lookups for the SIP signaling then click “Next”:

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Step 4: Enter the FQDN or IP Address(s) of the Unified Messaging Server(s) in the Network and then click “Next”:

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Step 5: Configure the T1 or E1 interface that will connect to the PBX by selecting the proper settings from the applicable drop down boxes and then click “Finish”:

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5.1. TLS Setup
Step 1: Configure the General Settings in VX Builder to include the Certificate Name that will be used for the MTLS communications and enable “Allow untrusted root certificates” (The Certificate Name must be the FQDN of the VX node):

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Step 2: Configure the SIP Trunk Group settings to enable “Mutual TLS” and “Allow SIP URI in TLS”:

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Step 3: Generate and receive MTLS Certificate: 1. Login to the VX node using Telnet or SSH and issue “generate certificate request” from an enabled level 2. Complete the steps and ensure the Common Name is the FQDN of the VX Node, entered in the VX Builder General section from Step 1 above 3. Open an FTP session to the VX node, and navigate to the ‘Certificates’ directory 4. Get the certificate that was generated above Step 4: Sign Certificate: Using the Certificate Signing procedures applicable to the Domain that the VX is in, submit the certificate to be signed, using base 64 encoded PKCS #10 or PKCS #7 file format. Download the base 64 encoded Certificate Chain. Save this file as “filename.p7b” type format. Step 5: Install Signed Certificate on VX Node: 1. Open an FTP session to the VX node, and navigate to the ‘Certificates’ directory 2. Put the “filename.p7b” signed certificate into the ‘Certificates’ directory 3. Login to the VX node using Telnet or SSH and issue “import certificate filename.p7b” from an enabled level, with filename.p7b being the signed certificate name 4. Ensure the signed certificate is installed on the VX node by issuing the “show certificate” command from and enabled level, with the certificate showing in the output

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5.2. SRTP Setup
Step 1: Insert SRTP key into VX Builder ‘Crypto Media Class’ section. Ensure the ‘Crypto Suite’ drop down selects ‘AES_CM_128_HMAC_SHA1_80’, and the rest of the config items as below:

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Step 2: Enable SRTP on the SIP Trunk Group: Select ‘Terminate SRTP Only’ from the Media Handling drop down box and select the proper Media Crypto Class configured in Step 1 above:

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6. PBX Setup Notes
Information specific to the VX and PBX UM Server Integration UMS Pilot Extension – 2562 Extensions enabled for Microsoft Exchange Server 2007 UM – 5100 - 5199

6.1. Create Routing for OCS Extensions and Exchange UM
Step 1: Change Telephony Settings The settings below must be made correctly for the calls to redirect to the OCS. ? Delay Time and delay count are 5 and 4 respectively. ? Companding is ALAW or ULAW based on your system configuration. ? Allow Outgoing transfer is CRITICAL. If not checked there can be no transfer to the OCS.

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Step 2: Set the PRI Line This example is a E1 ISDN QSIG B. ? Select CRC as desired. Must match the VX configuration or the leased line to the VX. ? CRITICAL – The Line Sub Type MUST be QSIG B for correct operation. ? In this case the TEI is 0 ? In this example all channel types are set for 30. ? For the PBX Clock is typically Network and Line Signalling (Side) is CPE.

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Step 3: Change Short Codes A Short Code is how the IP Office outputs calls. ? Set as noted in example below.

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Step 4: Change Incoming Call Route Set per the example below.

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6.2. TLS Setup
? N/A.

6.3. SRTP Setup
? N/A.

6.4. Fail-Over Configuration
? N/A.

6.5. Tested Phones
Meridian Digital Phone Sets

6.6. Other Comments
N/A

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7. Exchange 2007 UM Validation Test Matrix
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) 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.

Refer to:

Part I Mandatory Scenarios

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 user’s mailbox. Confirm hearing the prompt: “Welcome, you are connected to Microsoft Exchange. To access your mailbox, enter your extension…”

(P/CP/F/NT/NA)

Reason for Failure (see 6.1 for more detailed descriptions)

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P

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Navigate mailbox using the Voice User Interface (VUI). Navigate mailbox using the Telephony User Interface (TUI). Dial user extension and leave a voicemail. Dial user extension and leave a voicemail from an internal extension. Confirm the Active Directory name of the calling party is displayed in the sender field of the voicemail message.

P

3

P

4 4a

P

4b

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

P

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Confirm the correct phone number of the Pcalling party is displayed in the sender field of the voicemail message. 5 Dial Auto Attendant (AA). Dial the extension for the AA and confirm the AA answers the call. 6 6a Call Transfer by Directory Search. Call Transfer by Directory Search and have the called party answer. Confirm the correct called party answers the phone. 6b Call Transfer by Directory Search when the called party’s phone is busy. Confirm the call is routed to the called party’s voicemail. 6c Call Transfer by Directory Search when the called party does not answer. Confirm the call is routed to the called party’s voicemail. 6d Setup an invalid extension number for a particular user. Call Transfer by Directory Search to this user. Confirm the number is reported as invalid. 7 Outlook Web Access Phone Feature. (OWA) Play-OnP P P P P

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Listen to voicemail using OWA’s Play-OnPhone feature to a user’s extension. Listen to voicemail using OWA’s Play-OnPhone feature to an external number. 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: “Welcome, you are connected to Microsoft

P

7b

P

8

P

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Exchange. <User>. Please enter your pin and press the pound key.” 9 Send a test extension. FAX message to user P

Confirm the FAX is received in the user’s inbox. 10 Execute Test-UMConnectivity.

P

Part II Additional Scenarios

No.

Call Scenarios (see appendix for more detailed instructions) Setup TLS between gateway/IP-PBX and Exchange UM. Replace this italicized text with your TLS configuration: self-signed certificates or Windows Certificate Authority (CA).

(P/CP/F/NT/NA)

Reason for Failure (see 6.1 for more detailed descriptions)

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11a

Dial the pilot number and logon to a user’s mailbox. Confirm UM answers the call and confirm UM responds to DTMF input.

P

11b

Dial a user extension and leave a voicemail. Confirm the user receives the voicemail.

P

11c

Send a test FAX extension.

message

to

user

P

Confirm the FAX is received in the user’s inbox. 12 Setup TLS and SRTP between gateway/IP-PBX and Exchange UM. Replace this italicized text with your TLS configuration: self-signed certificates or Windows Certificate Authority (CA).

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12a

Dial the pilot number and logon to a user’s mailbox. Confirm UM answers the call and confirm UM responds to DTMF input.

P

12b

Dial a user extension and leave a voicemail. Confirm the user receives the voicemail.

P

12c

Send a test FAX extension.

message

to

user

P

Confirm the FAX is received in the user’s inbox. 13 Setup G.723.1 on the gateway. (If already using G.723.1, setup G.711 A Law or G.711 Mu Law for this step). Dial the pilot number and confirm the UM system answers the call. 14 Setup Message Waiting Indicator (MWI). Geomant offers a third party solution: MWI 2007. Installation files and product documentation can be found on Geomant’s MWI 2007 website. 15 Applicable only to IP-PBXs: Setup and test fail-over configuration on the IPPBX to work with two UM servers. Applicable only to IP-PBXs: Setup and test configuration involving transfer between multiple phone endpoints connected to IP-PBX wherein the transfer target is unconditionally forwarded to UM and the IP-PBX acts as a Back-to-Back User Agent. NA P P

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NA

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7.1. Detailed Description of Limitations

None

8. Troubleshooting

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Appendix
1. Dial Pilot Number and Mailbox Login
? ? ? ? Dial the pilot number of the UM server from an extension that is NOT enabled for UM. Confirm hearing the greeting prompt: “Welcome, you are connected to Microsoft Exchange. To access your mailbox, enter your extension...” Enter the extension, followed by the mailbox PIN of an UM-enabled user. Confirm successful logon to the user’s mailbox.

2. Navigate Mailbox using Voice User Interface (VUI)
? ? ? ? Logon to a user’s UM mailbox. If the user preference has been set to DTMF tones, activate the Voice User Interface (VUI) under personal options. Navigate through the mailbox and try out various voice commands to confirm that the VUI is working properly. This test confirms that the RTP is flowing in both directions and speech recognition is working properly.

3. Navigate Mailbox using Telephony User Interface (TUI)
? ? ? ? Logon to a user’s UM mailbox. If the user preference has been set to voice, press “#0” to activate the Telephony User Interface (TUI). Navigate through the mailbox and try out the various key commands to confirm that the TUI is working properly. This test confirms that both the voice RTP and DTMF RTP (RFC 2833) are flowing in both directions.

4. Dial User Extension and Leave Voicemail
? Note: If you are having difficulty reaching the user’s UM voicemail, verify that the coverage path for the UM-enabled user’s phone is set to the pilot number of the UM server.

a. From an Internal Extension
? ? ? From an internal extension, dial the extension for a UM-enabled user and leave a voicemail message. Confirm the voicemail message arrives in the called user’s inbox. Confirm this message displays a valid Active Directory name as the sender of this voicemail.

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b. From an External Phone
? ? ? From an external phone, dial the extension for a UM-enabled user and leave a voicemail message. Confirm the voicemail message arrives in the called user’s inbox. Confirm this message displays the phone number as the sender of this voicemail.

5. Dial Auto Attendant(AA)
? Create an Auto Attendant using the Exchange Management Console: ? ? ? ? ? ? ? ? Under the Exchange Management Console, expand “Organizational Configuration” and then click on “Unified Messaging”. Go to the Auto Attendant tab under the results pane. Click on the “New Auto Attendant…” under the action pane to invoke the AA wizard. Associate the AA with the appropriate dial plan and assign an extension for the AA. Create PBX dialing rules to always forward calls for the AA extension to the UM server. Confirm the AA extension is displayed in the diversion information of the SIP Invite.

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

6. Call Transfer by Directory Search
? Method one: Pilot Number Access ? ? ? Dial the pilot number for the UM server from a phone that is NOT enabled for UM. To search for a user by name: Press # to be transferred to name Directory Search. ? ? Call Transfer by Directory Search by entering the name of a user in the same Dial Plan using the telephone keypad, last name first.

To search for a user by email alias: ? ? ? Press “#” to be transferred to name Directory Search Press “# #” to be transferred to email alias Directory Search Call Transfer by Directory Search by entering the email alias of a user in the same Dial Plan using the telephone keypad, last name first.

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Method two: Auto Attendant ? ? Follow the instructions in appendix section 5 to setup the AA. Call Transfer by Directory Search by speaking the name of a user in the same Dial Plan. If the AA is not speech enabled, type in the name using the telephone keypad.

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?

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 or email alias.

a. Called Party Answers
? ? Call Transfer by Directory Search to a user in the same dial plan and have the called party answer. Confirm the call is transferred successfully.

b. Called Party is Busy
? ? Call Transfer by Directory Search to a user in the same dial plan when the called party is busy. Confirm the calling user is routed to the correct voicemail.

c. Called Party does not Answer
? ? Call Transfer by Directory Search to a user in the same dial plan and have the called party not answer the call. Confirm the calling user is routed to the correct voicemail.

d. The Extension is Invalid
? Assign an invalid extension to a user in the same dial plan. An invalid extension has the same number of digits as the user’s dial plan and has not been mapped on the PBX to any user or device. ? ? ? ? ? UM Enable a user by invoking the “Enable-UMMailbox” wizard. Assign an unused extension to the user. Do not map the extension on the PBX to any user or device. Call Transfer by Directory Search to this user. Confirm the call fails and the caller is prompted with appropriate messages.

7. Play-On-Phone
? To access play-on-phone: ? ? ? ? Logon to Outlook Web Access (OWA) by going to URL https://<server name>/owa. After receiving a voicemail in the OWA inbox, open this voicemail message. At the top of this message, look for the Play-On-Phone field ( Click this field to access the Play-On-Phone feature. Play on Phone...).

a. To an Internal Extension
? ? ? Dial the extension for a UM-enabled user and leave a voicemail message. Logon to this called user’s mailbox in OWA. Once it is received in the user’s inbox, use OWA’s Play-On-Phone to dial an internal extension.

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?

Confirm the voicemail is delivered to the correct internal extension.

b. To an External Phone number
? ? ? ? ? ? Dial the extension for a UM-enabled user and leave a voicemail message. Logon to the UM-enabled user’s mailbox in OWA. Confirm the voicemail is received in the user’s mailbox. Use OWA’s Play-On-Phone to dial an external phone number. Confirm the voicemail is delivered to the correct external phone number. Troubleshooting: ? Make sure the appropriate UMMailboxPolicy dialing rule is configured to make this call. As an example, open an Exchange Management Shell and type in the following commands: $dp = get-umdialplan -id <dial plan ID> $dp.ConfiguredInCountryOrRegionGroups.Clear() $dp.ConfiguredInCountryOrRegionGroups.Add("anywhere,*,*,") $dp.AllowedInCountryOrRegionGroups.Clear() $dp.AllowedInCountryOrRegionGroups.Add(“anywhere") $dp|set-umdialplan $mp = get-ummailboxpolicy -id <mailbox policy ID> $mp.AllowedInCountryGroups.Clear() $mp.AllowedInCountryGroups.Add("anywhere") $mp|set-ummailboxpolicy The user must be enabled for external dialing on the PBX. Depending on how the PBX is configured, you may need to prepend the trunk access code (e.g. 9) to the external phone number.

? ? ? ? ? ? ? ? ? ? ? ?

8. Voicemail Button
? ? ? ? Configure a button on the phone of a UM-enabled user to route the user to the pilot number of the UM server. Press this voicemail button on the phone of an UM-enabled user. Confirm you are sent to the prompt: “Welcome, you are connected to Microsoft Exchange. <User Name>. Please enter your pin and press the pound key.” 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.

9. FAX

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? ?

Use the Management Console or the Management Shell to FAX-enable a user. Management Console: ? ? ? Double click on a user’s mailbox and go to Mailbox Features tab. Click Unified Messaging and then click the properties button. Check the box “Allow faxes to be received”.

?

Management Shell - execute the following command: ? Set-UMMailbox –identity UMUser –FaxEnabled:$true

?

To test fax functionality: ? ? ? Dial the extension for this fax-enabled UM user from a fax machine. Confirm the fax message is received in the user’s inbox. Note: You may notice that the UM server answers the call as though it is a voice call (i.e. you will hear: “Please leave a message for…”). When the UM server detects the fax CNG tones, it switches into fax receiving mode, and the voice prompts terminate. Note: UM only support T.38 for sending fax.

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10.Test-UMConnectivity
? ? ? Run the Test-UMConnectivity diagnostic cmdlet by executing the following command in Exchange Management Shell: Test-UMConnectivity –UMIPGateway:<Gateway> -Phone:<Phone> |fl <Gateway> is the name (or IP address) of the gateway which is connected to UM, and through which you want to check the connectivity to the UM server. Make sure the gateway is configured to route calls to UM. <Phone> is a valid UM extension. First, try using the UM pilot number for the hunt-group linked to the gateway. Next, try using a CFNA number configured for the gateway. Please ensure that a user or an AA is present on the UM server with that number. The output shows the latency and reports if it was successful or there were any errors.

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11. MUTUAL TRANSPORT SECURITY LAYER (MTLS)
? ? Setup TLS on the gateway/IP-PBX and Exchange 2007 UM. Import/Export all the appropriate certificates.

a. Dial Pilot Number and Mailbox Login
? Execute the steps in scenario 1 (above) with TLS turned on.

b. Dial User Extension and Leave a Voicemail
? Execute the steps in scenario 4 (above) with TLS turned on.

c. FAX

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?

Execute the steps in scenario 9 (above) with TLS turned on.

12.MUTUAL TRANSPORT SECURITY LAYER (MTLS) AND SECURE RTP (SRTP)
? ? Setup TLS and SRTP on the gateway/IP-PBX and Exchange 2007 UM. Import/Export all the appropriate certificates.

a. Dial Pilot Number and Mailbox Login
? Execute the steps in scenario 1 (above) with TLS and SRTP turned on.

b. Dial User Extension and Leave a Voicemail
? Execute the steps in scenario 4 (above) with TLS and SRTP turned on.

c. FAX
? Execute the steps in scenario 9 (above) with TLS turned on.

13.G.723.1
? ? ? ? Configure the gateway to use the G.723.1 codec for sending audio to the UM server. If already using G.723.1 for the previous set of tests, use this step to test G.711 A Law or G.711 Mu Law instead. Call the pilot number and verify the UM server answers the call. Note: If the gateway is configured to use multiple codecs, the UM server, by default, will use the G.723.1 codec if it is available.

14.Message Waiting Indicator (MWI)
? ? Although Exchange 2007 UM does not natively support MWI, Geomant has created a 3rd party solution - MWI2007. This product also supports SMS message notification. Installation files and product documentation can be found on Geomant’s MWI 2007 website.

15.Test Fail-Over Configuration on IP-PBX with Two UM Servers
? This is only required for direct SIP integration with IP-PBXs. If the IP-PBX supports fail-over configuration (e.g., round-robin calls between two or more UM servers): ? ? ? ? Provide the configuration steps in Section 5. Configure the IP-PBX to work with two UM servers. Simulate a failure in one UM server. Confirm the IP-PBX transfers new calls to the other UM server successfully.

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16.Transfer between multiple phone endpoints connected to IP-PBX
? This is only required for direct SIP integration with IP-PBXs. If there are scenarios wherein multiple phone endpoints are connected to IP-PBX and the IP-PBX acts as a Back-to-Back User Agent (B2BUA) between these endpoints and Exchange UM then: ? ? ? ? Setup three phone endpoints (A, B and C) Configure C to forward all calls to UM unconditionally Call from A to B and then transfer the call to C Confirm that A is able to leave a message for C

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