Abstract. Testing was conducted via the Internal Interoperability Program at the Avaya Solution and Interoperability Test Lab.

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Interoperability Testing of SIP based Avaya Survivable Modular Messaging Solution using Avaya Message Storage Server with Avaya Aura TM Communication Manager and Avaya Aura TM Session Manager using Acronis True Image Server Issue 1.0 Abstract These Application Notes describe the configuration steps required to integrate SIP based Avaya Survivable Modular Messaging using Avaya Message Storage Server (MSS) with Avaya Aura TM Communication Manager and Avaya Aura TM Session Manager. It includes the procedure for building the solution, failover of the voic service to the Secondary Site following a full site outage at the Primary Site, and recovery back to the Primary Site. It also includes the procedure for building a single master WebLM server which will be used by both the Primary and Secondary site. The Modular Messaging systems are configured for MultiSite operation. Testing was conducted via the Internal Interoperability Program at the Avaya Solution and Interoperability Test Lab. 1 of 70

2 1. Introduction These Application Notes describe the steps required to configure Avaya Survivable Modular Messaging solution for Avaya Message Storage Server using Avaya Modular Messaging 5.2. The failover capabilities of Avaya Aura TM Communication Manager or Avaya Aura TM Session Manager will not be validated in these application notes. The procedure defined for building the solution, the procedure for failover of the voic service to the Secondary Site following a full site outage at the Primary Site and the procedure for recovery back to the Primary Site were also validated. In order to create an exact copy of the Primary Avaya MAS at the Secondary Site, an image copy is made of the Primary Avaya MAS and sent to the Secondary Site for restoration using Acronis True Image Server Version 10. Modular Messaging systems are configured for MultiSite operation. Particular attention will be paid to licensing and how each Avaya Messaging Application Server is licensed using a single master WebLM server which will be used by both the Primary and Secondary site. These Application Notes do not describe how to install a single master WebLM server, installation details can be found in reference [9]. SIP endpoints only were tested and these were registered to Avaya Aura TM Communication Manager. The test configuration in Figure 1 consisted of three separate locations. Site A contained Avaya Aura TM Communication Manager Feature Server, Avaya Aura TM Session Manager, DNS server, FTP server and WebLM server. Site A was connected, via a SIP trunk, to the Primary Avaya Modular Messaging system at Site B (Primary Site). During normal operations, the S8720 Server with G650 Media Gateway at Site A and Avaya Modular Messaging Server at Site B handled all call and voic needs for all subscribers. The Avaya Modular Messaging Servers at Site C (Secondary Site) were on standby during normal operations and did not handle calls or voic s for subscribers. Figure 1: Survivable Avaya Modular Messaging Network Configuration 2 of 70

3 2. Equipment and Software Validated The following equipment and software versions were used for the sample configuration provided in these Application Notes. Equipment Avaya S8720 Media Server Avaya G650 Media Gateway TN799DP C-LAN Circuit Pack TN2312BP IP Server Interface TN2602 IP Media Pro TN2224CP Digital Line Avaya S8510 Server with SM-100 card Software Avaya Aura TM Communication Manager (R015x ) HW01 FW034 HW215 FW047 HW08 FW049 HW08 FW015 Avaya Aura TM Session Manager Avaya S8510 Server Avaya Aura TM System Manager Avaya Modular Messaging Messaging Storage Server S8730 Avaya Modular Messaging 5.2 with patch Avaya Modular Messaging Messaging Application Server S8730 Avaya Modular Messaging 5.2 ( ) with patch Avaya 9620 IP Telephone (SIP) Avaya 9630 IP Telephone (SIP) Acronis True Image Server Acronis Version 10 WebLM server WebLM release 4.5.5, Microsoft Windows 2003 server Table 1: Hardware and Software Versions 3 of 70

4 3. Configure Avaya Aura TM Communication Manager This section shows the configuration in Communication Manager. All configurations in this section are administered using the System Access Terminal (SAT). These Application Notes assumed that the basic configuration has already been administered. For further information on Communication Manager, please consult with references [1] and [2] The procedures include the following areas: Verify Avaya Aura TM Communication Manager License Administer System Parameters Features Administer IP Node Names Administer IP Network Region Administer Codec set Administer SIP Signaling Group Administer Trunk Group Administer Route Pattern Administer Private Numbering Administer Dial Plan Administer AAR analysis Administer Feature Access Codes Administer Hunt Group Administer Coverage Path Administer SIP Phone Save Changes 4 of 70

5 3.1. Verify Avaya Aura TM Communication Manager License Use the display system-parameter customer options command to verify the Maximum Administered SIP Trunks field value with the corresponding value in the USED column. The difference between the two values needs to be greater than or equal to the desired number of simultaneous SIP trunk connections. Note: The license file installed on the system controls the maximum capacity permitted. If there is insufficient capacity or a required feature is not enabled, contact an authorized Avaya sales representative to make the appropriate changes. display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 30 0 Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: 0 0 Max Concur Registered Unauthenticated H.323 Stations: 0 0 Maximum Video Capable Stations: 10 1 Maximum Video Capable IP Softphones: 10 4 Maximum Administered SIP Trunks: Administer System Parameters Features Use the change system-parameters features command to allow for trunk-to-trunk transfers. This feature is needed to allow for transferring an incoming/outgoing call from/to a remote switch back out to the same or different switch. For simplicity, the Trunk-to-Trunk Transfer field was set to all to enable all trunk-to-trunk transfers on a system wide basis. Note: This feature poses significant security risk and must be used with caution. As an alternative, the trunk-to-trunk feature can be implemented using Class Of Restriction or Class Of Service levels. change system-parameters features Page 1 of 18 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? y Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 10 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? y Music/Tone on Hold: none Music (or Silence) on Transferred Trunk Calls? no DID/Tie/ISDN/SIP Intercept Treatment: attd Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred Automatic Circuit Assurance (ACA) Enabled? n 5 of 70

6 3.3. Administer IP Node Names Use the change node-names ip command to add entries for the Communication Manager and Session Manager that will be used for connectivity. In the sample network, clan and are entered as Name and IP Address for the CLAN card in Communication Manager running on the Avaya S8720 Server. In addition, SM100 and are entered for Session Manager. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address EnterpriseCM Gateway SM SM clan default medpro procr Administer IP Network Region Use the change ip-network-region n command, where n is the network region number to configure the network region being used. In the sample network ip-network-region 1 is used. For the Authoritative Domain field, enter the SIP domain name configured for this enterprise and a descriptive Name for this ip-network-region. Set Intra-region IP-IP Direct Audio and Inter-region IP-IP Direct Audio to yes to allow for direct media between endpoints. Set the Codec Set to 1 to use ip-codec-set 1, see the following section. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: Authoritative Domain: silstack.com Name: StackFeatureCM MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 6 of 70

7 3.5. Administer Codec Set Use the change ip-codec-set n command, where n is an available ip-codec-set number as defined in Section 3.4, to select the audio codec type supported for calls to Modular Messaging. change ip-codec-set 1 Page 1 of 2 Codec Set: 1 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n Administer SIP Signaling Group In the test configuration, Communication Manager acts as a Feature Server. An IMS enabled SIP trunk is required to reach the Session Manager. Use the add signalinggroup n command, where n is the signaling-group number being added to the system. Set IMS Enabled to y. Use the values defined in Section 3.3 and 3.4 for Near-end Node Name, Far-End Node-Name and Far-End Network Region. The Far-end Domain is left blank so that the signaling group accepts any authoritative domain. add signaling-group 150 Page 1 of 2 SIGNALING GROUP Group Number: 150 IMS Enabled? y IP Video? n Group Type: sip Transport Method: tcp Near-end Node Name: clan Far-end Node Name: SM100 Near-end Listen Port: 5065 Far-end Listen Port: 5065 Far-end Network Region: 1 Far-end Domain: Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? y Direct IP-IP Early Media? n H.323 Station Outgoing Direct Media? y Alternate Route Timer(sec): 6 7 of 70

8 3.7. Administer SIP Trunk Group Use the add trunk-group n command, where n is the new trunk group number being added to the system. The following screens show the settings used for trunk group 150. add trunk-group 150 Page 1 of 21 TRUNK GROUP Group Number: 150 Group Type: sip CDR Reports: y Group Name: To ASM SM100 COR: 1 TN: 1 TAC: 150 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Navigate to Page 3 and enter private for Numbering Format. Signaling Group: 150 Number of Members: 10 add trunk-group 150 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: private UUI Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Show ANSWERED BY on Display? Y Navigate to Page 4 and enter 101 for Telephone Event Payload Type. add trunk-group 150 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? n Network Call Redirection? n Send Diversion Header? n Support Request History? y Telephone Event Payload Type: of 70

9 3.8. Administer Route Pattern Configure a route pattern to correspond to the newly added SIP trunk group. Use the change route-pattern n command, where n is the route pattern number specified in Section Configure this route pattern to route calls to trunk group number 150 configured in Section 3.7. Assign the lowest FRL (facility restriction level) to allow all callers to use this route pattern. change route-pattern 150 Page 1 of 3 Pattern Number: 150 Pattern Name: To ASM SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: n user 2: n user 3: n user 4: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR M 4 W Request Dgts Format Subaddress 1: y y y y y n n rest none 2: y y y y y n n rest none 3: y y y y y n n rest none 4: y y y y y n n rest none 3.9. Administer Private Numbering Use the change private-numbering command to define the calling party number to be sent out through the SIP trunk. In the sample network configuration below, all calls originating from a 5-digit extension beginning with 34 will result in a 5-digit calling number. The calling party number will be in the SIP From header. change private-numbering 0 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: of 70

10 3.10. Administer Dial Plan Use the change dialplan analysis command to define Dialed String 34 of Total Length 5 as an ext Call Type. change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 3 dac 2 5 aar 34 5 ext 4 5 aar 8 1 fac 9 1 fac * 3 fac # 3 fac Administer AAR analysis Use the change aar analysis n command where n is the dial string pattern to configure an aar entry for Dialed String 34666, the MM pilot number, to cover calls to the voic using Route Pattern 150. change aar analysis 34 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 2 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd aar n aar n aar n aar n aar n Administer Feature Access Code Configure a feature access code to use for AAR routing. Use the change feature access code command to define an Auto Alternate Routing (AAR) Access Code. change feature-access-codes Page 1 of 8 FEATURE ACCESS CODE (FAC) Abbreviated Dialing List1 Access Code: Abbreviated Dialing List2 Access Code: Abbreviated Dialing List3 Access Code: Abbreviated Dial - Prgm Group List Access Code: Announcement Access Code: Answer Back Access Code: Attendant Access Code: Auto Alternate Routing (AAR) Access Code: *8 Auto Route Selection (ARS) - Access Code 1: Access Code 2: Automatic Callback Activation: Deactivation: Call Forwarding Activation Busy/DA: *1 All: Deactivation: Call Forwarding Enhanced Status: Act: Deactivation: Call Park Access Code: Call Pickup Access Code: CAS Remote Hold/Answer Hold-Unhold Access Code: CDR Account Code Access Code: Change COR Access Code: Change Coverage Access Code: Conditional Call Extend Activation: Deactivation: 10 of 70

11 3.13. Administer Hunt Group Configure a Hunt Group to be used as the call coverage point for the call coverage path assigned to MAS subscribers. Use the add hunt-group n command where n is the hunt group number to be assigned. Configure a Group Name and Group Extension number to be used as the Modular Messaging pilot name and number. Select ucd-mia for Group Type. add hunt-group 5 Page 1 of 60 HUNT GROUP Group Number: 5 Group Name: mmvoice Group Extension: Group Type: ucd-mia TN: 1 COR: 1 Security Code: ISDN/SIP Caller Display:mbr-name ACD? n Queue? n Vector? n Coverage Path: Night Service Destination: MM Early Answer? n Local Agent Preference? n Navigate to Page 2. Select sip-adjunct for Message Center. For Voice Mail Number and Voice Mail Handle use the Group Extension number and Group Name defined on Page 1 respectively. For Routing Digits use the AAR access code defined in Section add hunt-group 5 Page 2 of 60 HUNT GROUP Message Center: sip-adjunct Voice Mail Number Voice Mail Handle Routing Digits (e.g., AAR/ARS Access Code) mmvoice *8 11 of 70

12 3.14. Administer Coverage Path Configure a coverage path for the MAS subscribers. Use command add coverage path n where n is the coverage path number to be assigned. Configure a Coverage Point, using the hunt group value hn where n is the hunt group number, configured in Section add coverage path 5 COVERAGE PATH Coverage Path Number: 5 Cvg Enabled for VDN Route-To Party? n Next Path Number: Hunt after Coverage? n Linkage COVERAGE CRITERIA Station/Group Status Inside Call Outside Call Active? n n Busy? y y Don't Answer? y y Number of Rings: 2 All? n n DND/SAC/Goto Cover? y y Holiday Coverage? n n COVERAGE POINTS Terminate to Coverage Pts. with Bridged Appearances? n Point1: h5 Rng: Point2: Point3: Point4: Point5: Point6: Administer SIP Phone Modify the SIP Phone registered in Section 4.12 to allow call coverage to Modular Messaging. Use the command change station and on Page 1 set Coverage Path 1 to the coverage path defined in Section change station Page 1 of 6 STATION Extension: Lock Messages? n BCC: 0 Type: 9620SIP Security Code: TN: 1 Port: S00004 Coverage Path 1: 5 COR: 1 Name: 34205, Mick Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Time of Day Lock Table: Loss Group: 19 Message Lamp Ext: Display Language: english Button Modules: 0 Survivable COR: internal Survivable Trunk Dest? y IP SoftPhone? n IP Video? n 12 of 70

13 Navigate to Page 2 and set MWI Served User Type to sip-adjunct which is the message center defined in Section change station Page 2 of 6 STATION FEATURE OPTIONS LWC Reception: spe LWC Activation? y Coverage Msg Retrieval? y CDR Privacy? n Bridged Call Alerting? n Active Station Ringing: single Auto Answer: none Data Restriction? n Idle Appearance Preference? n Bridged Idle Line Preference? n H.320 Conversion? n Per Station CPN - Send Calling Number? y EC500 State: enabled MWI Served User Type: sip-adjunct Emergency Location Ext: Coverage After Forwarding? s Direct IP-IP Audio Connections? y Always Use? n IP Audio Hairpinning? n Administer Off PBX Telephone Station Mapping This section shows the off-pbx-telephone station-mapping. The SIP telephone extension uses off pbx Application OPS which is used for SIP enabled telephones. The SIP Trunk Selection is set to aar. The Config Set which is the desired call treatment was set to 1. display off-pbx-telephone station-mapping Page 1 of 3 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Application Dial CC Phone Number Trunk Config Dual Extension Prefix Selection Set Mode OPS aar Save Changes Use the save translation command to save all changes. save translation SAVE TRANSLATION Command Completion Status Error Code Success 0 13 of 70

14 4. Configuring Avaya Aura TM Session Manager This section provides the procedures for configuring Session Manager. For further information on Session Manager, please consult with references [3], [4] and [5]. The procedures include the following areas: Login to Avaya Aura TM System Manager Administer SIP domain Administer Adaptations Administer SIP Entities Administer Entity Links Administer Time Ranges Administer Routing Policies Administer Dial Patterns Administer Regular Expression Administer Avaya Aura TM Session Manager Add Avaya Aura TM Communication Manager as a Feature Server Add Users for SIP Phones 4.1. Log in to Avaya Aura TM System Manager Access the System Manager using a Web Browser and entering where <ip-address> is the IP address of System Manager. Log in using appropriate credentials and accept the subsequent Copyright Legal Notice. 14 of 70

15 By selecting Network Routing Policy from the left panel menu, a short procedure for configuring Network Routing Policy is shown on the right panel. 15 of 70

16 4.2. Administer SIP Domain Add the SIP authoritative domain for the communications infrastructure by selecting Network Routing Policy SIP Domains on the left panel menu and clicking the New button to create a new SIP domain entry. Complete the following options: Name The authoritative domain name (e.g., silstack.com) Notes Description for the domain (optional) Click Commit (not shown) to save changes. Verify the domain is created as in screenshot below. Note: Since the sample network does not deal with any foreign domains, no additional SIP Domains entry is needed. 16 of 70

17 4.3. Administer Adaptations In the sample configuration, multi-site Avaya Modular Messaging represents its subscribers using 12 digit telephone numbers. The 5 digit extensions used by the Avaya Communication Manager system are preceded by the 7 digits DigitConversionAdapter is used in Session Manager to convert between the 5 and 12 digit formats when routing between Modular Messaging and those systems. To add an Adaptation select Network Routing Policy Adaptations on the left panel menu and clicking the New button (not shown) to create a new Adaptation entry. For the Modular Messaging adaptation, enter the following information: Under General Adaptation Name An informative name for the adaptation e.g. Modular Messaging Adaption_MD Module Name DigitConversionAdapter Module Parameter The domain name as configured in Section 4.2, (e.g., Silstack.com) The remaining fields can be left blank. Under Digit Conversion for Incoming Calls to SM and Digit Conversion for Outgoing Calls from SM, click Add and then edit the fields in the resulting new row as shown below: Matching Pattern A Regular expression or partial digit string used to match the incoming or outgoing dialed number Min Minimum dialed number length Max Maximum dialed number length Delete Digits Number of digits to delete from the beginning Insert Digits The digit string to insert at the beginning Address to Modify Chose between origination, destination or both Click Commit (not shown) to save. 17 of 70

18 Incoming calls (SIP INVITE messages) from Modular Messaging that use 12 digit numbers will be converted to the 5 digit form by deleting the first 7 digits. Session Manager will route the call based on the resulting 5 digit extension. Outgoing calls from Session Manager that use 5 digit numbers that begin with 34 will have appended to the number. 18 of 70

19 4.4. Administer SIP Entities A SIP Entity must be added for Session Manager for each SIP-based telephony system supported by a SIP Trunk. To add a SIP Entity, select Network Routing Policy SIP Entities on the left panel menu and then click on the New button (not shown). Enter the following for each SIP Entity: Under General: Name An informative name (e.g., SessionManager) FQDN or IP Address IP address of the signaling interface on the Session Manager Type Session Manager for Session Manager Time Zone Time zone for this location Under Port, click Add, and then edit the fields in the resulting new rows Port Enter Port number used in Section 4.5 Protocol Transport protocol to be used to send SIP requests Click Commit to save changes. 19 of 70

20 The following screen shows the SIP Entity for Communication Manager. 20 of 70

21 The following screen shows the SIP Entity for Modular Messaging. The Adaptation field was configured with the Adaptation Module specified in Section 4.3. Note: In the event of an outage at the Primary Avaya Modular Messaging site the SIP Entity on the Secondary Site requires to be configured on the ModularMessaging_MD entity by configuring the FQDN or IP Address of the Secondary site. 21 of 70

22 4.5. Administer Entity Links A SIP trunk between a Session Manager and a telephony system is described by an Entity Link. To add an Entity Link, select Network Routing Policy Entity Links on the left panel menu and click on the New button. Fill in the following fields in the new rows that are displayed. Name An informative name SIP Entity 1 Select SessionManager Protocol Transport protocol to be used to send SIP requests Port Port number to which the other system sends its SIP requests SIP Entity 2 The other SIP Entity for this link, created in Section 4.4 Port Port number to which the other system expects to receive SIP requests Trusted Whether to trust the other system Click Commit to save changes. The following screen shows the Entity Links used in the sample network. 22 of 70

23 4.6. Administer Time Ranges Before adding routing policies (see next step), time ranges must be defined during which the policies will be active. In the sample network, one policy was defined that would allow routing to occur at anytime. To add this time range, select Network Routing Policy Time Ranges from the left panel menu and then click New on the right. Fill in the following fields. Name An informative name (e.g. Always) Mo through Su Check the box under each day of the week for inclusion Start Time Enter start time (e.g. 00:00 for start of day) End Time Enter end time (e.g. 23:59 for end of day) Click Commit to save changes. The following screen shows the Time Range configured in the sample network. 23 of 70

24 4.7. Administer Routing Policies A routing policy must be created to direct how calls will be routed to a system. One routing policy must be added for Modular Messaging. To add a routing policy, select Network Routing Policy Routing Policies on the left panel menu and then click on the New button (not shown). Enter the following for each Routing Policy: Under General Name An informative name (e.g., ModularMessaging_MD) Under SIP Entity as Destination, click Select, and then select the appropriate SIP entity to which this routing policy applies Under Time of Day, click Add, and then select the time range configured in the previous step. Click Commit to save changes. The following screen shows the Routing Policy Details for Modular Messaging. 24 of 70

25 4.8. Administer Dial Patterns A dial pattern must be defined that will direct calls to the appropriate telephony system. In the sample network, the 5-digit extension will be used as the pilot number for Modular Messaging. For pilot number configuration, perform the following. Select Network Routing Policy Dial Patterns on the left panel menu and then click on the New button (not shown). Under General Pattern Pilot number i.e Min Minimum length of the pilot number i.e. 5 Max Maximum length of the pilot number i.e. 5 Notes Comment on purpose of dial pattern SIP Domain Select ALL Navigate to Originating Locations and Routing Policies and select Add. 25 of 70

26 Under Originating Location select all locations by checking the box next to ALL and under Routing Policies select the Routing Policy created in Section 4.7. Click Select button to confirm the chosen options and return to the Dial Pattern screen (shown above). Click Commit button to save. 26 of 70

27 4.9. Administer Regular Expression Regular Expression must be defined for Avaya MAS subscribers so that they can communicate with Modular Messaging via the Session Manager. The format of the Regular Expression is where voic -handle is the handle defined in Section Select Network Routing Policy Regular Expressions on the left panel menu and then click on the New button (not shown). Under General Pattern Rank Order A unique number Notes Comment on purpose of Regular Expression Navigate to Routing Policy and select Add. Select the routing policy created in Section 4.7. Click Select button to confirm the chosen option and then return to the Regular Expression screen (shown above). Click Commit button as shown in the screen above to save. 27 of 70

28 4.10. Administer Avaya Aura TM Session Manager To complete the configuration, adding the Session Manager will provide the linkage between System Manager and Session Manager. Select Session Manager Session Manager Administration on the left. Then click Add (not shown) and fill in the fields as described below and shown in the following screen: Under General: SIP Entity Name Select the name of the SIP Entity added for Session Manager Description Descriptive comment (optional) Management Access Point Host Name/IP Enter the IP address of the Session Manager management interface Under Security Module Network Mask Default Gateway: Enter the network mask corresponding to the IP address of Session Manager Enter the IP address of the default gateway for Session Manager Use default values for the remaining fields. Click Commit to add this Session Manager. 28 of 70

29 4.11. Add Avaya Aura TM Communication Manager as a Feature Server In order for Communication Manager to provide configuration and Feature Server support to SIP phones when they register to Session Manager, Communication Manager must be added as an application Create an Application Entity Select Applications Entities on the left. Click on New (not shown). Enter the following fields and use defaults for the remaining fields: Name A descriptive name i.e. feature Type Select CM Node Enter the IP address for CM SAT access Navigate to the Attributes section and enter the following: Login Login used for SAT access Password Password used for SAT access Confirm Password Password used for SAT access Click on Commit to save. 29 of 70

30 Create a Feature Server Application Select Session Manger Application Configuration Applications on the left. Click on New (not shown). Enter following fields and use defaults for the remaining fields: Name A descriptive name SIP Entity Select the CM Application Entity defined in Section Click on Commit to save. 30 of 70

31 Create a Feature Server Application Sequence Select Session Manager Application Configuration Application Sequences on the left. Click on New (not shown). Enter a descriptive name in the Name field. Click on the + sign next to the appropriate Available Applications and they will move up to the Applications in this Sequence section. Click on Commit to save. 31 of 70

32 Synchronize Avaya Aura TM Communication Manager Data Select Communications System Management Telephony on the left. Select the appropriate Element Name from the list. Check the Initialize data for selected devices box. Then click on Now. This may take some time. Use the menus on the left under Monitoring Scheduler to determine when the task is complete. 32 of 70

33 4.12. Add Users for SIP Phones Users must be added via Session Manager and the details will be updated on the CM. Select User Management User Management on the left. Then click on New (not shown). Enter a Last Name and First Name. 33 of 70

34 Navigate to the Identity section and enter the following and use defaults for other fields: Login Name The desired where domain was defined in Section 4.2 Password Password for user to log into SMGR Shared Communication Profile Password Password to be entered by the user when logging into the phone 34 of 70

35 Navigate to and click on Communication Profile section to expand that section, use the default values. Then click on Communication Address to expand that section, click New and enter the following: Type Select SIP SubType Select username Fully Qualified Address Enter the Click on Add. 35 of 70

36 Navigate to and click on Session Manager section to expand. Select the appropriate Session Manager server for Session Manager Instance. For Origination Application Sequence and Termination Application Sequence select the application sequence created in Section Click on Station Profile to expand that section. Enter the following fields and use defaults for the remaining fields: System Select the CM Entity created in Section Extension Enter a desired extension number Template Select a telephone type template Port Select IP Click on Commit to save (not shown). 36 of 70

37 5. Install and Configure Avaya Messaging Application Server This section focuses on the installation and configuration procedures for Avaya MAS to support Survivable Modular Messaging and the required configuration changes upon a disaster at the Main site to ensure correct functionality of the Avaya MAS at the Disaster Recover site for voic support. The configuration procedures include the following areas: Configuration of Primary Modular Messaging Site Configuration of Secondary Modular Messaging Site Steps to maintain service during an outage. The Data Collection Tool (DCT) file is created as part of pre-installation planning and used as input to automate MAS installations. Full details of configuring a Modular Messaging system using a DCT file are described in reference [7] Configure Primary Avaya Modular Messaging Site This section deals with the configuration of the Primary Avaya Modular Messaging Site. It also describes how to make a copy of the Primary Site installation that can be used to create the Secondary Site installation. It is assumed that Avaya MAS and Avaya MSS servers have the correct software installed and are appropriately licensed. Ensure that the FQDN of Avaya MSS and Avaya MAS below are configured on the DNS servers to point to Primary Avaya MAS and Avaya MSS. The IP addresses of all Avaya Modular Messaging Servers are static. MAS masdh1.silstack.com MSS mssdh1.silstack.com Note: Private Windows domain is used for communication between Avaya MSS and Avaya MAS MultiSite Configuration In order to enable MultiSite functionality on the Primary Modular Messaging Site the MAS must be configured as shown in this section. The configuration can be verified by following the instructions in this section. Log in to the Avaya MAS server using the appropriate credentials. Select Start Programs Avaya Modular Messaging Voice Mail System Configuration to start the Voice Mail System Configuration tool. 37 of 70

38 From the Voice Mail System Configuration window go to Voice Mail Domains SMMLAB Sites. From the subsequent Voice Mail Domain window, confirm that the Enable MultiSite checkbox is selected in the screen as shown. Click Configure to open the Site Configuration window. In this example the Dublin site has an ID of , a Full Mailbox length of 12 and a Short Mailbox length of 5. The PBX Name is the name of the PBX as shown in Voice Mail Domains PBXs described later in this section. 38 of 70

39 To configure or confirm the SIP integration of Avaya MAS, from the Voice Mail System Configuration window, go to Voice Mail Domains SMMLAB PBXs and click on the SIP tab, click on the + icon and add the Session Manager s IP address under Address/FQDN, TCP for Protocol, and click the MWI box so message waiting notifications will be sent. Configure Protocol and SRTP settings to match the SM settings discussed in Section 4.5. In this example, SRTP was not enabled and the chosen Protocol was TCP. Enter the SIP Domain to match the Gateway Name configured in Section 4.2. Click the Configure button located near the bottom of the screen to configure the incoming and outgoing phone number translation rules. 39 of 70

40 The Translation Rules screen opens as shown below. The example translation rules shown below were added by selecting Add and will map an incoming number of to its canonical form of Proper operation of the rules can be verified by adding Test inputs in the left pane and viewing the resulting output in the corresponding rule in the right pane. Click on OK when finished, then again on OK in the original Add new PBX window (see previous screen). For more details in creating incoming and outgoing translation rules please consult with reference [7]. 40 of 70

41 Configuration of MWI and Call Me Services From the Voice Mail System Configuration Window, go to Voice Mail Domains SMMLAB and double click Message Waiting Indicator. In the Message Waiting Indicator screen that opens, check Enable Message Waiting Indicator (MWI). Then set Message Application Servers that support MWI to the hostname of the Primary Site MAS server as shown. Click OK. 41 of 70

42 From the Voice Mail System Configuration Window, go to Voice Mail Domains SMMLAB and double click Call Me. In the Call Me popup that appears, check Enable Call Me and set MAS Call Me Server as shown. 42 of 70

43 Configuration of Services This section describes how to configure an example Class-of-Service that can be used in creating Avaya Modular Messaging subscribers. Configuration is performed through the Modular Messaging Message Administration application. To launch the application, enter Avaya MSS hostname or IP address in the URL field of a web browser. Log in with the appropriate credentials. The following webpage is displayed. Select the Classes-of-Service menu option from the Messaging Administration Menu on the left of the screen. The Manage Classes-of-Service screen opens up. Select an unused class of service from the displayed list and click Edit the Selected COS as shown. 43 of 70

44 The Edit a Class-of-Service screen opens. Scroll down the screen to the Subscriber Feature and Services section and set the options as appropriate for the configuration. In this case Call Me Allowed, Find Me Allowed and Message Waiting Indication Allowed are all set to yes. The remaining settings can be left at their default values. Save any changes made by clicking Save which is located at the bottom of the screen (not shown). 44 of 70

45 Subscriber Creation Click on the Subscriber Management option in the Messaging Administration Menu. Enter the Local Subscriber Mailbox Number and click Add or Edit. Here the Mailbox Number is The Edit Local Subscriber screen opens. In this example, the Canonical form of the PBX Extension is used. This refers to the combination of the Site ID, as configured in Section 5.1.1, and the Switch Native extension, in this case In the interoperability testing, the same telephone extensions for the subscribers were used for the Mailbox Number, Numeric Address, and PBX Extension fields. Enter in the appropriate details for the subscriber and ensure that the Class-of-Service is set to the one described in Section The default Community ID is selected. Scroll down to the bottom of the screen and click Save (not shown). Repeat this section to add all subscribers. 45 of 70

46 5.2. Single Master WebLM Server From Modular Messaging 5.2, WebLM manages the Modular Messaging licences and the same licences cannot be shared by both the Primary and Secondary Modular Messaging systems. A survivable Modular Messaging license system was installed using a single master WebLM server Access the WebLM Interface The browser-based GUI of WebLM is accessed using the URL where <ip-address> is the IP address of the WebLM server. Log on with the appropriate credentials. 46 of 70

47 View Licensed Products To view the number of licenses available select Licensed Products MODULAR_MESSAGING_5_2 Modular_Messaging_5_2 from the left pane. This will show the number of Acquired Licences. 47 of 70

48 Configuring WebLM URL on Modular Messaging System To configure the WebLM server on the MAS in the Primary site, launch the Voice Mail System Configuration (VMSC) software application by selecting Start Programs Avaya Modular Messaging Voice Mail System Configuration. From the Voice Mail System Configuration window, select Voice Mail Domains SMMLAB Licensing and set the WebLM URL field to where <ip-address> is that of the master WebLM server. Wait for the system to acquire the required licenses and to update with the particular license mode status. It can take up to 10 minutes for the license mode to change. The License Mode should show as Normal on the Primary MAS. 48 of 70

49 To configure the WebLM server on the Secondary MAS edit the WebLM URL field and enter in an invalid WebLM server URL, e.g. the system will show the License Mode as Error. 49 of 70

50 5.3. Backup and Restoration of MAS and MSS In order to create an exact copy of the Primary Avaya MAS at the Secondary Site, an image copy is made of the Primary Avaya MAS and sent to the Secondary Site for restoration. In this case Acronis True Image Server version 10 is used to copy the image of Avaya MAS onto a USB II hard drive with at least 300GB capacity. Full details of how to use Acronis True Image Server Software are available in reference [10] and will not be described here. A bootable rescue media is created using Acronis True Image Server that has been installed on a server. A copy of this media should be made available at both the Primary and Secondary Sites. Place this disk in the CD tray of the Primary Avaya MAS server and reboot. Once booted, the Acronis software asks the user which option to run, select Acronis Backup & Recovery of 70

51 Backup Avaya MAS Select the Back up option to start the Backup wizard of the Avaya MAS. When configuring the Backup operation, select the items to back up, in this example Disk 1 (C:), as shown below, and where to back up, in this example E:\Name:5.2_MAS_Primary_S8730, as shown below. To start the back-up process click Ok (not shown). Once the backup is complete the 300GB hard drive can be shipped to the Secondary Avaya Modular Messaging Site and used as described in Section of 70

52 Backup Avaya Messaging Storage Server Data This section describes the steps involved in creating a backup of all data stored on the Primary Avaya MSS sever. This backup is made to an FTP server that can by reached via the network from the Primary Avaya MSS. Using the Modular Messaging Message Administration application introduced in Section 5.1.3, click the Remote Storage Configuration option on the Backup/Restore Menu. Set the following parameters and click Save: Backup Mode Select FTP Host Name Set this to point to a working FTP server User Name A user name on the FTP server Directory The directory path on the FTP server where to store files Password The associated password for the user 52 of 70

53 Select Backup from the Backup/Restore Menu and ensure System Data, Names and Greetings and Messages are set to yes. Then click Start Backup. Stop Voice System is optional while performing the backup. If the Voice System is stopped prior to the backup, remember to restart it from the Utilities Menu (not shown) by clicking Start Messaging when backup is complete. 53 of 70

54 Restore Avaya Messaging Storage Server This section and section deal with the restoration of Primary Site Avaya MSS data collected in Section and the Primary Site Avaya MAS image created in Section to the Secondary Modular Messaging Site. Launch the Messaging Administration tool on the Secondary Avaya MSS as described in Section Ensure that the Secondary Avaya MSS server can connect to the FTP server that contains the backup files created in Section and configure the Remote Storage Configuration appropriately. Backup Mode Select FTP. Host Name Set this to point to a working FTP server. User Name A user name on the FTP server. Directory The directory path on the FTP server where the files are stored. Password The associated password for the user. 54 of 70

55 Select Restore from the Backup/Restore Menu and click Stop Voice System. When the messaging system is stopped click Restore again. The following screen is displayed. Select the file that was created in Section and ensure System Data, Names and Greetings and Messages are set to yes. Then click Start Restore. 55 of 70

56 Once the restore operation is completed, the system prompts the user to reboot. Before this can happen the TCP/IP Network Configuration should be reset to the values required for operation at the Secondary Site. This is because the restore operation overwrites the values with those of the Primary Site. Click on TCP/IP Network Configuration in the Server Administration Menu and change the Default Gateway Address and IP Address fields to reflect the values needed at the Secondary Site. Click Save (not shown) Default Gateway Address Changed from to IP Address Changed from to Reboot Avaya MSS by clicking Reboot Server from the Utilities Menu and clicking Reboot (not shown) in the subsequent screen. In normal operation, it is also recommended to restart Avaya MAS after a restoration of MSS data. 56 of 70

57 Restore Avaya Messaging Application Server Place the bootable rescue media from Acronis created in Section 5.3 into the CD tray of the Secondary Avaya MAS server. Connect the USB II hard drive created in Section and reboot. Once booted, the Acronis software asks the user which option to run, select Acronis Backup & Recovery 10. The following screen is displayed. Click Recover to start the Recovery Wizard. 57 of 70

58 Select the date to recover which was backed up from Section To start the recovery process click OK (not shown). 58 of 70

59 Post Avaya Messaging Application Server Restoration Configuration Once the recovery process is complete the IP address details of the Secondary Avaya MAS must be changed to reflect the settings of the Secondary Avaya MAS and not those of the Primary system. To do this log into Avaya MAS with the appropriate credentials and click Start Settings Network Settings. Select Local Area Connection and then click Properties from the subsequent dialog. Select Internet Protocol (TCP/IP) and open its Properties dialog. Ensure that the IP address and Default gateway are correct for the secondary site. In the example the IP address is changed from to and the Default gateway from to of 70

60 Click Advanced to open the Advanced TCP/IP Settings dialog and ensure that Register this connection s addresses in DNS field under the DNS tab is unchecked as shown. 60 of 70

61 Update the hosts file located in C:\WINDOWS\system32\drivers\etc directory to ensure the correct IP addresses are used for both Avaya Modular Messaging Servers at the Secondary Site. Reboot Avaya MAS. Log in to the Secondary Avaya MAS with the appropriate credentials and open a Windows Explorer. Locate the C:\Avaya_Support\Tools\FEDBSync directory and run the FEDBSync Tool. The tool is shown below. Click Execute. 61 of 70

62 Click OK to continue. Wait for event 1027 to appear in the Windows Event Viewer. Once completed, subscriber functionality will operate correctly. 62 of 70

63 6. Avaya Modular Messaging Configuration upon Outage In the event of an outage at the Primary Avaya Modular Messaging site, call coverage will not be available on the Secondary site until the following actions are complete. Change the FQDN of Avaya MSS and Avaya MAS on the DNS servers in the enterprise to point to the IP addresses of the secondary Avaya MAS and Avaya MSS. Details of how to configure a DNS server are outside the scope of these Application Notes. Note: In the event of an outage at the Primary Avaya Modular Messaging site the SIP Entity on the Secondary Site requires to be configured on the ModularMessaging_MD entity by configuring the FQDN or IP Address of the Secondary site, described in Section 4.4. Log in to the Secondary Avaya MAS and check the Register this connection s addresses in DNS field under the DNS tab of the Advanced TCP/IP Settings described in Section on Avaya MAS. In order to ensure the Message Waiting Indicator (MWI) is correctly reflected on subscriber handsets, an MWI update can be initiated using the Voice Mail System Configuration tool. To launch the Voice Mail System Configuration (VMSC) software application select Start Programs Avaya Modular Messaging Voice Mail System Configuration. Select Voice Mail Domains SMMLAB Message Waiting Indicator Start MWI Update. 63 of 70

64 The Start On-Demand MWI Update screen opens. Select the Re-evaluate mailbox MWI rules for lamp state (slower but more accurate) radio button to force an accurate update of individual subscriber lamp states. Click OK. 64 of 70

65 7. Verification Verification can be carried out in several points in the test configuration. They include: Verify that users can dial from one subscriber to another and the call covers to voice mail. Verify that the proper greeting is played. Verify when the Primary Modular Messaging server disconnects a user can place a call from one subscriber to another and the call covers to the voice mail of the Recovery Modular Messaging. Leave a voice message and verify that the MWI of the user s physical phone is lit. Verify that when failover occurs to the Secondary Modular Messaging server that the MWI of the users physical phone is lit with the messages that were on the Primary Modular Messaging server. Delete the voice messages and verify that the MWI lamp is extinguished. Verify that the SIP Entity links created for the Communication Manager and Modular Messaging in the Session Manager are in an Up status. Verify that the SIP trunk is in-service using the status trunk command on the Communication Manager. Verify that the SIP signaling group is in-service using the status signaling-group command on the Communication Manager Verify Call Covers to Voice Mail at Primary Site Make a call from one subscriber to another and verify that the call covers to Modular Messaging upon no answer. Leave a message for the subscriber. Verify the MWI lamp on the phone is lit where the message was left. From the subscriber that received the call, dial the Modular Messaging Pilot number to retrieve the message. Verify that the Modular Messaging system identifies the subscriber and that the voice message can be retrieved. Verify that once the message has been retrieved the MWI lamp extinguished Verification Call Covers to Voice Mail at Secondary Site Disconnect the Primary Modular Messaging Servers and follow the procedure outlined in Section 5 on how to modify configuration upon outage. Place a call to the pilot number from the same subscribers phone used in this section. Ensure that the correct Welcome message for that subscriber is played. This indicates that the caller is recognized by the voic system. Compare the subscriber options associated with that subscriber with those noted down in the previous section. The following Modular Messaging features were verified to work properly with the MAS in the Main site before the failover and properly with the MAS in the Disaster Recovery site after the failover. Call Me Find Me Message Waiting Indicator 65 of 70

66 7.3. Verify SIP Entity Links are in a UP Status on System Manager Select Session Manager System Status SIP Entity Monitoring. Verify as shown below that none of the SIP entities created in Section 4.4 links are down, indicating that they are all reachable for routing. 66 of 70

67 Entity Link Status Click on the SIP Entity Names AvayaCM and ModularMessaging_MD, shown in the previous screen, and verify that the connection status is Up, as shown in the screen shots below. 67 of 70