Configuring Avaya Aura Communication Manager Messaging 6.0 with Avaya Aura Session Manager 6.0 Issue 1.0

Transcription

1 Avaya Solution & Interoperability Test Lab Configuring Avaya Aura Communication Manager Messaging 6.0 with Avaya Aura Session Manager 6.0 Issue 1.0 These Application Notes present a sample configuration for a network that connects Avaya Aura Communication Manager Messaging 6.0 to Avaya Aura Session Manager 6.0 using SIP trunks. For this sample configuration, Avaya Aura Communication Manager Messaging 6.0 runs on an Avaya S8510 Server with Avaya G450 Media Gateway, and Avaya Aura Session Manager runs on an Avaya S8510 Server. Avaya Aura Communication Manager Messaging is a feature that integrates voice, Fax, text messages, and attachments into a single system and offers subscribers enhanced flexibility to manage messages from their telephones or personal computers. 1 of 41

2 Contents Contents Introduction Equipment and Software Validated Configure Avaya Aura Communication Manager as Evolution Server Verify Communication Manager License Configure Dial Plan analysis Configure IP Node Names Configure IP Codec Sets Configure IP Network Region Administer a SIP Trunk to Session Manager Add IP interface Add SIP Signaling group Configure a SIP Trunk Group Configure Route Pattern Configure Public Unknown Numbering Administer AAR Analysis Add Hunt Group Change feature-access-code Add Coverage path Add Stations Save Translations Configuration for Avaya Aura Session Manager Specify SIP Domain Add Locations Add SIP Entities Add Entity Links Add Time Ranges Add Routing Policies Add Dial Patterns Add Regular expression Add Session Manager Configure Communication Manager Messaging Server Configuring Switch Admin Link Configuring Messaging Configuring Switch Administration Start Messaging Messaging server Admin Configure voic service for Communication Manager Subscribers Verification Steps Communication Manager Verification Communication Manager Messaging configuration verification SIP Monitoring on Session Manager Verification Scenarios of 41

3 7. Conclusion Additional References of 41

4 MESSAGE OK CONTACTS MENU CALL LOG SPEAKER MUTE VOLUME HEADSET ABC DEF GHI JKL MNO PQRS TUV WXYZ * Avaya one-x [ 0 # OK CONTACTS MENU CALL LOG.,@ ABC DEF GHI JKL MNO PQRS TUV WXYZ * Avaya one-x SPEAKER MUTE VOLUME HEADSET [ 0 # PHONE PHONE ME SS AGE OK CONTACTS MENU CALL LOG.,@ ABC DEF GHI JKL MNO PQRS TUV WXYZ * Avaya one-x SPEAKER MUTE VOLUME HEADSET ME SS AGE [ 0 # OK CONTACTS MENU CALL LOG.,@ ABC DEF GHI JKL MNO PQRS TUV WXYZ * Avaya one-x SPEAKER MUTE VOLUME HEADSET [ 0 # PHONE PHONE 1. Introduction These Application Notes present a sample configuration for a network that connects Avaya Aura Communication Manager Messaging 6.0 and Avaya Aura Session Manager using SIP trunks. In the sample configuration, H.323 telephones were registered with Avaya Aura Communication Manager integrated with Messaging, forming a single box solution called Avaya Aura Communication Manager Messaging. Avaya Aura Communication Manager Messaging supports two types of integration H.323 and SIP. These application notes will only discuss the SIP integration. All inter-system calls are carried over these SIP trunks. Communication Manager Messaging provides a voic facility to the user. When a call is not answered or the call is busy a voice message can be recorded. The sample configuration is shown in Figure 1. Communication Manager Messaging runs on an Avaya S8510 Server with Avaya G450 Media Gateway. Avaya Aura Session Manager runs on an Avaya S8510 Server. Avaya Aura Communication Manager Messaging embedded application is packaged as part of the solution template that is used to install Avaya Aura Communication Manager. Figure 1 Test Configuration used in these Application Notes 4 of 41

5 The setup has configuration has the following entities. Avaya Aura Communication Manager (Evolution Server) Communication Manager is connected to Session Manager using a SIP trunk. It manages routing of calls for the endpoints connected. It also re-routes calls to Communication Manager Messaging via Session manager on call no-answer or busy. Avaya Aura Communication Manager Messaging. Communication Manager messaging provides a voic facility for the H.323 and SIP endpoints. Communication Manager works as an Evolution Server and is configured with H.323 stations. SIP stations are configured on Session Manager and are supported by Communication Manager working as a Feature Server. Subscribers and extension data is configured on Communication Manager Messaging for creating voic box. Note: Communication Manager Messaging and Communication Manager both are co-resident on the same box, but do not communicate directly to each other. The communication is over SIP trunk via Session Manager. Avaya Aura Session Manager The main function of Session Manager is to route calls from Communication Manager intended for Communication Manager Messaging via SIP trunks. Avaya Aura System Manager System Manager provides support for configuration of a Session Manager using a web interface. Avaya G450 Media Gateway The G450 Media Gateway is configured with Communication Manager Evolution Server and Communication Manager as Feature Server if present. When the call involving a non-ip station is set up successfully, the media is sent through Media Gateway. Avaya H.323 phones Avaya 9640 IP Telephones are configured as H.323 stations. These phones register with Communication Manager. Avaya SIP phones Avaya 9640 IP Telephones are configured as SIP stations. SIP stations are registered and subscribe to Session Manager and have feature data configured on a separate Communication Manager configured as a Feature Server. Avaya Aura Communication Manager (Feature Server) This server is not required for H.323 phones. The messaging configuration on Communication Manager Feature Server is similar to Evolution Server. Section 3 discusses in detail configuration of Communication Manager as an Evolution Server. For simplicity these Application Notes discuss configuration with H.323 phones. For more details refer to the appropriate documentation for SIP station configuration in the references. The subsequent sections discuss configuration of each of these entities in detail. The figure shows two SIP links indicated by orange dashed lines. A CM-SM SIP link is created between Communication Manager and Session Manager on default port A second Communication 5 of 41

6 Manager Messaging-SM link is created between Communication Manager Messaging and Session Manager on port These Application Notes will focus on the configuration of the SIP trunks and call routing. Detailed administration of the endpoint telephones will not be described. SIP endpoints and the integration with Session Manager are configured as described in documents available from references. 2. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Product / Hardware Platform Avaya Aura TM Communication Manager Messaging Avaya S8510 Server Version Version: 6.0 SP build: CM build: CMM build: Session Manager 6.0 Avaya Aura TM Session Manager Avaya S8510 Server Avaya Aura TM System Manager System Manager 6.0 Avaya S8510 Server Avaya G450 Media Gateway Avaya IP Telephones: 9640 (SIP) version (H.323) version S3.110b 3. Configure Avaya Aura Communication Manager as Evolution Server. This section provides the procedures for configuring Communication Manager. The procedures include the following areas: Verify Communication Manager license Dial Plan analysis Configure IP node names Verify/List IP interfaces Configure Codec Set Configure Network Region Administer a SIP Trunk to Session Manager Configure Route Pattern Configure Location and Public Unknown Numbering Administer AAR Analysis Add hunt group Add coverage path Add station(s) Save Translations 6 of 41

7 Throughout this section the administration of Communication Manager is performed using a System Access Terminal (SAT). The following commands are entered on the system with the appropriate administrative permissions. Some administration screens have been abbreviated for clarity. These instructions assume that the Communication Manager has been installed, configured, licensed and provided with a functional dial plan. Refer to the appropriate documentation as described in references for more details Verify Communication Manager License Use the display system-parameters customer-options command. Navigate to Page 2 and verify that there is sufficient remaining capacity for SIP trunks by comparing 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. Verify the highlighted value, as shown below. display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP econs: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: Maximum Administered Ad-hoc Video Conferencing Ports: Maximum Number of DS1 Boards with Echo Cancellation: Maximum TN2501 VAL Boards: Maximum Media Gateway VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: (NOTE: You must logoff & login to effect the permission changes.) If there is insufficient capacity of SIP Trunks or a required feature is not enabled, contact an authorized Avaya sales representative to make the appropriate changes. 7 of 41

8 3.2. Configure Dial Plan analysis In these Application Notes the system was configured with a 5 digit dialplan. As shown below, dialed strings that begin with 2 with a total length of 5 are assigned to extension numbers. Dialed strings that begin with 3 with a total length of 5 will be routed to Communication Manager Messaging, using AAR tables. Dialplan can be verified with the display dialplan analysis command. Add following entries to the table. Note extensions used are in the range and So there are two entries with call type ext. display dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 3 dac 2 5 ext 3 5 ext 8 1 fac 9 1 fac * 3 fac # 3 fac 3.3. Configure IP Node Names All SIP signaling with Session Manager is carried through an IP interface. When configuring a SIP Trunk in Communication Manager, use the IP address of the Session Manager s asset card. Use the change node-names ip command to add the Name and IP Address for the Session Manager, in the example asm and change node-names ip Page 1 of 2 Name IP Address asm default procr procr6 :: IP NODE NAMES Note that in the example some other values (default, procr, and procr6) have been already created as per installation and configuration of Communication Manager Configure IP Codec Sets Use the command change ip-codec-set n command where n is the codec set used in the configuration. Enter the following values: Audio Codec Set for G.711MU/ G.711A needed to support Communication Manager Messaging. Silence Suppression: Retain the default value n. Frames Per Pkt: Enter 2. Packet Size (ms): Enter 20. Retain the default values for the remaining fields, and submit these changes. 8 of 41

9 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 : G.711A n Configure IP Network Region Use the change ip-network-region n command, where n is the number of the network region used and set the Intra-region IP-IP Direct Audio, and Inter-region IP-IP Direct Audio fields to yes. For the Codec Set enter the corresponding audio codec set configured in Section 3.4. Set the Authoritative Domain to the SIP domain. Retain the default values for the remaining fields, and submit these changes. Note: In the test configuration, network region 1 was used. When creating a new network region or modifying another one, ensure to configure it with the correct parameters. change ip-network-region 1 Page 1 of 20 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: cmm.silpunelab.com Name: cmmregion 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: Administer a SIP Trunk to Session Manager Add IP interface Use the add ip-interface procr command to define an interface on Communication Manager to communicate with Session Manager. Enable Interface: y Network Region: 1 add ip-interface procr Page 1 of 2 IP INTERFACES Type: PROCR Target socket load: Enable Interface? y Allow H.323 Endpoints? y Allow H.248 Gateways? y Network Region: 1 Gatekeeper Priority: 5 Node Name: procr Subnet Mask: /24 IPV4 PARAMETERS IP Address: 9 of 41

10 Add SIP Signaling group Use the add signaling-group n command, where n is an available signaling group number, for one of the SIP trunks to Session Manager, and fill in the indicated fields. Default values can be used for the remaining fields: Group Type: sip Transport Method: tcp Peer Detection Enabled?: n Peer Server: SM Near-end Node Name: procr Far-end Node Name: Session Manager node name from Section 3.3 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 1 Far-end Domain: cmm.silpunelab.com (same as domain from Section 4.1) DTMF over IP: rtp-payload IMS Enabled?: n This is for Communication Manager Evolution Server. Note: For configuration of Communication Manager as Feature Server the following change is required. IMS Enabled?: y add signaling-group 1 Page 1 of 1 SIGNALING GROUP Group Number: 1 Group Type: sip IMS Enabled? n Transport Method: tcp Q-SIP? n IP Video? n Peer Detection Enabled? n Peer Server: SM SIP Enabled LSP? n Enforce SIPS URI for SRTP? y Near-end Node Name: procr Far-end Node Name: asm Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 1 Far-end Domain: cmm.silpunelab.com 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 Initial IP-IP Direct Media? y H.323 Station Outgoing Direct Media? Alternate Route Timer(sec): Configure a SIP Trunk Group Add the corresponding trunk group controlled by this signaling group via the add trunk-group n command, where n is an available trunk group number and fill in the indicated fields. 10 of 41

11 Group Type: sip Group Name: A descriptive name TAC: An available trunk access code i.e. 101 Service Type: tie Signaling Group: signaling group number added in Section i.e. 1 Number of Members: The number of SIP trunks to be allocated to calls routed to Session Manager (must be within the limits of the total trunks available from licensed verified in Section 3.1) Note: the number of members determines how many simultaneous calls can be processed by the trunk through Session Manager. add trunk-group 1 Page 1 of 21 TRUNK GROUP Group Number: 1 Group Type: sip CDR Reports: y Group Name: cmm-trunkgrp COR: 1 TN: 1 TAC: 101 Direction: two-way Outgoing Display? n Dial Access? n Queue Length: 0 Service Type: tie Night Service: Auth Code? n Member Assignment Method: auto Signaling Group: 1 Number of Members: 210 Navigate to page 3 and change Numbering Format to public. Use default values for all other fields. Submit these changes. add trunk-group 1 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public UUI Treatment: service-provider Go to page 4, and set Convert 180 to 183 for Early Media? to n. add trunk-group 1 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: Convert 180 to 183 for Early Media? n Always Use re-invite for Display Updates? n Enable Q-SIP? n 3.7. Configure Route Pattern Configure a route pattern to correspond to the newly added SIP trunk group. Use change route pattern n command, where n is an available route pattern. Enter the following values for the specified fields, and retain the default values for the remaining fields. Submit these changes. 11 of 41

12 Pattern Name: A descriptive name i.e. cmm-route Grp No: The trunk group number from Section FLR: Enter a level that allows access to this trunk, with 0 being least restrictive. change route-pattern 1 Page 1 of 3 Pattern Number: 1 Pattern Name: cmm-route 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: 1 0 n user 2: n user 3: n user 4: n user 5: n user 6: 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 rehu 3.8. Configure Public Unknown Numbering Use the change public-unknown-numbering 1 command, to define the calling party number to be sent to Communication Manager Messaging. Add an entry for the trunk group defined in Section In the example shown below, all calls originating from a 5-digit extension beginning with 2 and routed to trunk group 1 will result in a 5-digit calling number. The calling party number will be in the SIP From header. Submit these changes. For Communication Manager: Ext Len: Number of digits for extension. i.e. 5 Trk Grp: Trunk group number. i.e. 1 Ext Code: Enter range for cm extensions. i.e. 2 For Communication Manager Messaging: Ext Len: Number of digits for extension. i.e. 5 Trk Grp: Trunk group number. i.e. 1 Ext Code: Enter the Communication Manager Messaging number. i.e change public-unknown-numbering 1 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: Administer AAR Analysis This section provides sample Automatic Alternate Routing (AAR) used for routing calls with dialed digits 3xxxx to Communication Manager Messaging. Note that other methods of routing may be used. 12 of 41

13 Use the change aar analysis 0 command and add an entry to specify how to route the calls. Enter the following values for the specified fields and retain the default values for the remaining fields. Submit these changes. Dialed String: Dialed prefix digits to match on, in this case 2 and Total Min: Minimum number of digits, in this case 5 Total Max: Maximum number of digits, in this case 5 Route Pattern: The route pattern number from Section 3.7. i.e. 1 Call Type: aar change aar analysis 2 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd aar n aar n Add Hunt Group. Create new hunt group for Communication Manager Messaging. Enter the following details Group Name: Descriptive name for hunt-group to be configured. Group Extension: Communication Manager Messaging group extension i.e Coverage Path: Configure coverage path for Communication Manager Messaging. See section 3.12 Security code: Numbers 0-9 add hunt-group 1 Page 1 of 60 HUNT GROUP Group Number: 1 ACD? n Group Name: cmm-huntgrp Queue? n Group Extension: Vector? n Group Type: ucd-mia Coverage Path: 1 TN: 1 Night Service Destination: COR: 1 MM Early Answer? n Security Code: 1234 Local Agent Preference? n ISDN/SIP Caller Display: mbr-name On Page 2 set the fields as shown below. Message Center: sip-adjunct Voice Mail Handle: To match the first part of the regular expression created in SM. For example if regular expression in SM is punecmm@cmm.silpunelab.com, then enter the first part punecmm as the handle. AAR access code: 8 this number is used as the FAC. 13 of 41

14 add hunt-group 1 Page 2 of 60 HUNT GROUP Message Center: sip-adjunct Voice Mail Number Voice Mail Handle Routing Digits (e.g., AAR/ARS Access Code) punecmm Change feature-access-code. To enable 8 as the feature-access-code, execute change feature-access-codes command. Auto Alternate Routing (AAR) Access Code: 8 change feature-access-codes Page 1 of 10 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: 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: Contact Closure Open Code: Close Code: Add Coverage path Use the add coverage path n command, where n is an unassigned coverage path, to create a new coverage path with following values. Point1: Use h1 to indicate hunt-group 1 created in previous Section Rng: Number of rings after which hunt-group would be used. Set to value of 41

15 add coverage path 1 Page 1 of 1 COVERAGE PATH Coverage Path Number: 1 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: h1 Rng: 3 Point2: Add Stations This section provides a sample of stations being added to the configuration. These stations would be used to make calls. Use the command add station Station 1: Type: 9640 Security Code: 1234; A password is required when logging to station. Name: Name of the subscriber. Coverage path: The coverage path number section i.e. 1 add station Page 1 of 5 STATION Extension: Lock Messages? n BCC: 0 Type: 9640 Security Code: 1234 TN: 1 Port: IP Coverage Path 1: 1 COR: 1 Name: John Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Time of Day Lock Table: Loss Group: 19 Personalized Ringing Pattern: 1 Message Lamp Ext: Speakerphone: 2-way Mute Button Enabled? y Display Language: english Button Modules: 0 Survivable GK Node Name: Survivable COR: internal Survivable Trunk Dest? y Media Complex Ext: IP SoftPhone? n IP Video? n Short/Prefixed Registration Allowed: default Customizable Labels? y Station 2: Type: of 41

16 Security Code: 1234; A password is required when logging to station. Name: Name of the subscriber. Coverage path: The coverage path number section i.e. 1 add station Page 1 of 5 STATION Extension: Lock Messages? n BCC: 0 Type: 9640 Security Code: 1234 TN: 1 Port: IP Coverage Path 1: 1 COR: 1 Name: Dave Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Time of Day Lock Table: Loss Group: 19 Personalized Ringing Pattern: 1 Message Lamp Ext: Speakerphone: 2-way Mute Button Enabled? y Display Language: english Button Modules: 0 Survivable GK Node Name: Survivable COR: internal Media Complex Ext: Survivable Trunk Dest? y IP SoftPhone? n IP Video? n Short/Prefixed Registration Allowed: default Customizable Labels? y Save Translations Configuration of Communication Manager is complete. Use the save translations command to save these changes. 4. Configuration for Avaya Aura Session Manager This section provides the procedures for configuring Session Manager, assuming it has been installed and licensed as described in the references. The procedures include adding the following items: Specify SIP Domain. Logical/physical Locations that can be occupied by SIP Entities. SIP Entities corresponding to the SIP telephony systems and Session Manager. Entity Links, which define the SIP trunk parameters used by Session Manager when routing calls to/from SIP Entities. Time Ranges during which routing policies are active. Routing Policies, which control call routing between the SIP Entities. Dial Patterns, which govern to which SIP Entity a call is routed. Regular Expression, which is used for call routing to Communication Manager Messaging. Session Manager, corresponding to the Session Manager Server to be managed by System Manager. Configuration is accomplished by accessing the browser-based GUI of System Manager, using the URL where <ip-address> is the IP address of System Manager. 16 of 41

17 Log in to the system with valid credentials. The menu shown below is displayed. Expand the Routing link on the left side as shown. The sub-menus displayed in the left column below will be used to configure all but the last of the above items 4.1. Specify SIP Domain Add the SIP domain for which the communications infrastructure will be authoritative. Do this by selecting SIP Domains on the left and clicking the New button (not shown) on the right. The following screen will then be shown. Fill in the following fields and click Commit. Name: The authoritative domain name (e.g. cmm.silpunelab.com) Notes: Descriptive text (optional). 17 of 41

18 4.2. Add Locations Locations can be used to identify logical and/or physical locations where SIP Entities reside, for purposes of bandwidth management. Location is added to the configuration for Session Manager and Communication Manager Messaging. To add a location, select Locations on the left and click on the New button (not shown) on the right. The following screen will then be shown. Fill in the following: Under General: Name: Notes: Managed Bandwidth: Under Location Pattern: IP Address Pattern: Notes: A descriptive name for CMM. Descriptive text (optional). Use the default value. An ip-address pattern used to logically identify the location. Descriptive text (optional). The screen below shows addition of the lab location, which has a single Communication Manager Messaging location. Click Commit to save. 18 of 41

19 Add Location for Session Manager. Under General: Name: Notes: Managed Bandwidth: Under Location Pattern: IP Address Pattern: location. Notes: A descriptive name for Session Manager. Descriptive text (optional). Use the default value. An ip-address pattern used to logically identify the Descriptive text (optional). 19 of 41

20 4.3. Add SIP Entities A SIP Entity must be added for Session Manager and for each SIP-based telephony system supported by it using SIP trunks. In the sample configuration, the following SIP entities are added Session Manager(avaya-asmc), Communication Manager (punecmm-cm), and Communication Manager Messaging (punecmm). To add a SIP Entity, select SIP Entities on the left and click on the New button (not shown) on the right. The following screen is displayed. Fill in the following: Under General: Name: A descriptive name for Communication Manager. FQDN or IP Address: IP address of the signaling interface Type: CM for Communication Manager and CM for Communication Manager Messaging Location: Select one of the locations defined previously. i.e.: CMM Time Zone: Time zone for this location. 20 of 41

21 Defaults can be used for the remaining fields. Click Commit to save each SIP Entity definition. The following screen shows the addition of Communication Manager. The IP address used is that of the procr. Keep Adaptation as blank. Location is same for Communication Manager, Communication Manager Messaging, and Session Manager. The following screen shows the addition of Session Manager. The IP address used is that of the asset card. Name: FQDN or IP Address: Type: Location: Time Zone: A descriptive name for Session Manager. IP address of the ASM or the signaling interface Select Session Manager for Session Manager, Select one of the locations defined previously. i.e.: ASM-C Time zone for this location. 21 of 41

22 Under Entity Links, click Add, and then edit the fields in the resulting new row as shown below: Port: Protocol: Port number on which the system listens for SIP requests. Transport protocol to be used to send SIP requests. i.e. TCP The following screen shows the addition of Communication Manager Messaging. Note that both Communication Manager and Communication Manager Messaging share the same IP address, that of the Communication Manager procr. 22 of 41

23 4.4. Add Entity Links A SIP trunk between Session Manager and a telephony system is described by an Entity Link. To add an Entity Link, select Entity Links on the left and click on the New button (not shown) on the right. Fill in the following fields in the new row that is displayed: Name: A descriptive name. SIP Entity 1: Select the Session Manager entity. Port: Port number to which the other system sends SIP requests SIP Entity 2: Select the name of the other system. Port: Port number on which the other system receives SIP requests. These ports should match SIP signaling ports. Trusted: Check this box. Note: If this box is not checked, calls from the associated SIP Entity will be denied. Protocol: Select the transport protocol among UDP/TCP/TLS Check these are aligned with the definition on the other end of the link. In the example, TCP is used. Click Commit to save each Entity Link definition. The following screen illustrates adding the Entity Link for Communication Manager. 23 of 41

24 Below is illustrated adding the Entity Link for Communication Manager Messaging. 24 of 41

25 4.5. Add Time Ranges Before adding routing policies (see next section), time ranges must be defined during which the policies will be active. In the sample configuration, one policy was defined that would allow routing to occur at anytime. To add this time range, select Time Ranges: on the center of the Time Ranges page under the heading, click on the New button (not shown). Fill in the following: Name: A descriptive name (e.g., 24/7). Mo through Su Check the box under each of these headings Start Time Enter 00:00. End Time Enter 23:59 Click Commit to save this time range Add Routing Policies Routing policies describe the conditions under which calls will be routed to the SIP Entities specified in Section 4.3. Two routing policies must be added one for Communication Manager and one for Communication Manager Messaging. 25 of 41

26 To add a routing policy, select Routing Policies on the left and click on the New button (not shown) on the right. The following screen is displayed. Fill in the following: Under General: Name: Enter a descriptive name for the Communication Manager policy. Under SIP Entity as Destination: Click Select, and then select the appropriate SIP entity created for Communication Manager. Under Time of Day: Click Add, and select the time range configured in the previous section. Defaults can be used for the remaining fields. Click Commit to save the Routing Policy definition. The following screen shows the Routing Policy for Communication Manager. Under General: Name: Enter a descriptive name for Communication Manager Messaging policy. 26 of 41

27 Under SIP Entity as Destination: Click Select, and then select the appropriate SIP entity created for Communication Manager Messaging. Under Time of Day: Click Add, and select the time range configured in the previous section. Defaults can be used for the remaining fields. Click Commit to save the Routing Policy definition. Below is illustrated the Routing Policy for Communication Manager Messaging, configured in these sample application notes Add Dial Patterns Dial patterns must be defined that will direct calls to the appropriate SIP Entity. In the sample configuration, 5-digit extensions beginning with 20 reside on Communication Manager and 5- digit resides on Communication Manager Messaging. To add a dial pattern, select Dial Patterns on the left and click on the New button (not shown) on the right. Fill in the following, as shown in the screen below, which corresponds to the dial pattern for routing calls to Communication Manager: 27 of 41

28 Under General: Pattern: Dialed number or prefix. Min Minimum length of dialed number. Max Maximum length of dialed number. Notes Comment on purpose of dial pattern. Under Originating Locations and Routing Policies: Click Add, and then select the appropriate location and routing policy from the list. Default values can be used for the remaining fields. Click Commit to save this dial pattern. The following screen shows the dial pattern definitions for Communication Manager. The following screen shows the dial pattern definitions for Communication Manager Messaging. 28 of 41

29 4.8. Add Regular expression. Pattern: Note punecmm was configured in Communication Manager hunt-group messaging configuration as in section Rank Order: Usually this is the next number in the current rank order sequence. Routing Policy: Select the routing policy created for Communication Manager Messaging as section of 41

30 4.9. Add Session Manager To complete the configuration, adding the Session Manager will provide the linkage between System Manager and Session Manager. Expand the Session Manager Elements menu on the left and select Session Manager Administration. 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 the asset card interface i.e Enter the IP address of the default gateway for asset card interface i.e of 41

31 Use default values for the remaining fields. Click Save to add this Session Manager. Save the data and check the Session Manager configuration as shown below. 31 of 41

32 5. Configure Communication Manager Messaging Server This section describes the steps for configuring the Avaya Aura Communication Manager Messaging to interoperate with Avaya Aura Session Manager via SIP trunking. (see Figure 1). The following steps describe the administrative procedures for configuring Communication Manager Messaging: Configuring Switch Admin Link. Start Messaging. Messaging server Admin. Configuring Subscribers. Mapping Entries with Communication Manager Subscribers. The following instructions assume having logged on to the Communication Manager Messaging console using the web interface to access the GUI with the appropriate credentials Configuring Switch Admin Link Configuring Messaging Initially when a new system is installed, the Messaging feature is by default disabled on Communication Manager Messaging. To enable messaging, go to the Administration/Server Maintenance web page, and then click on to Messaging Software in Miscellaneous section as shown below. 32 of 41

33 Check the current state of Messaging is disabled and click the Enable button. Once the messaging is enabled the button name will change to Disable and the current state of messaging is updated to Internal Messaging is enabled. The Messaging option will now be available in the Administration tab as show below. Click on Messaging option Configuring Switch Administration Scroll down to the Switch Link Administration and click on Switch Link Admin; fill in the fields as described below and shown in the following screen: Extension length: As per configuration in Communication Manager. 5 digits Switch Integration type: SIP Transport Method: TCP Connection1: Enter SM asset card ip-address. Messaging Address: Ip-address same as for procr. 33 of 41

34 Port: Use value other than 5060, as this is used for CM-SM link and is configured in Communication Manager signaling configuration. (for example use 6060) SIP Domain Messaging: Use the same domain as configured ip-network-region Switch: Use same as Messaging domain. Click on save button Start Messaging To initialize the Messaging processes go to the Utilities section and click on Start Messaging. During initialization the software executes an audit database, and initializes voice cards. Finally when the process is up it displays the message Voice Messaging is now completely up 34 of 41

35 5.3. Messaging server Admin To configure the voic facility for a range of extensions, go to the Messaging server Admin page in the Server Administration section. Enter data for the following fields. Password: Confirm Password: Enter password for configuring Messaging server Re-enter data used in the previous field to confirm. Address ranges: Prefix: As per dial pattern enter the prefix for the extensions. Starting Extension: Enter the starting extension number. See section 3.8 Ending Extension: Enter the ending extension number. 35 of 41

36 5.4. Configure voic service for Communication Manager Subscribers. Add subscribers for which voic service is to be created. Browse to the Messaging Administration section and click on the Subscriber Management link. Create two subscribers with the same extension and Name as configured in Communication Manager. See section 3.13 Fill the data entries for each subscriber as shown below and save. Subscriber 1: Name: John Mailbox: Password: 1324 (Any numeric password) Subscriber 2: Name: Dave Mailbox: Password: 1324 (Any numeric password) 36 of 41

37 37 of 41

38 6. Verification Steps 6.1. Communication Manager Verification Verify the status of the signaling group is in-service, with the command status signaling-group n where n is the signaling group number configured in Section status signaling-group 1 STATUS SIGNALING GROUP Group ID: 20 Active NCA-TSC Count: 0 Group Type: sip Active CA-TSC Count: 0 Signaling Type: facility associated signaling Group State: in-service Verify the status of the trunk group is in-service/idle, using the command status trunk group n where n is the trunk group number configured in Section The following shows the trunk status with all lines idle for Communication Manager Messaging. status trunk 1 Page 1 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy 0001/001 T00001 in-service/idle no 0001/002 T00002 in-service/idle no 0001/003 T00003 in-service/idle no 0001/004 T00004 in-service/idle no 0001/005 T00005 in-service/idle no 0001/006 T00006 in-service/idle no 0001/007 T00007 in-service/idle no 0001/008 T00008 in-service/idle no 0001/009 T00009 in-service/idle no 0001/010 T00010 in-service/idle no 0001/011 T00011 in-service/idle no 0001/012 T00012 in-service/idle no 0001/013 T00013 in-service/idle no 0001/014 T00014 in-service/idle no 6.2. Communication Manager Messaging configuration verification SIP Monitoring on Session Manager Expand the Session Manager menu on the left and click on System Status, then on SIP Monitoring. Verify that none of the links to the defined SIP entities are down, indicating that they are all reachable for call routing. To check status of link entities configured in section 4.4, click on the link below. The status of the links should be UP as shown in the screenshot below. 38 of 41

39 6.3. Verification Scenarios Basic Call verification: Place a call from H.323 Station1 (ext 20001) to the H.323 Station2 (ext 20002) Do not answer Station After three rings the call goes to voic (Communication Manager Messaging). Station1 hears greeting from Communication Manager Messaging and prompts user to leave message. User records a voice message for called party, and presses # # to save. Message Waiting Indicator (MWI) on called party telephone is ON. User at Station2 sees MWI is ON. Dials Communication Manager Messaging number. User enters password and is able to listen to new message. After hearing voice message MWI is OFF. Other feature verification: User dials Communication Manager Messaging number. User is able record voice message and sends to other extensions. User can reply to voice messages received. User can forward voice message to other extensions. 39 of 41

40 7. Conclusion As illustrated in these Application Notes, Avaya Aura Communication Manager Messaging 6.0 SIP integration can successfully be configured with Avaya Aura Session Manager 6.0. The tests were carried out using SIP on TCP and media over RTP. However, for security reasons, it is recommended that SIP with TLS and media over SRTP be used. 8. Additional References Avaya references are available at Avaya Aura Communication Manager Messaging: Implementing Avaya Aura Communication Manager Messaging Doc ID: Avaya Aura Communication Manager Messaging Documentation CD Avaya Aura Communication Manager: Administering Avaya Aura Communication Manager Doc ID: Avaya Aura Session Manager & Avaya Aura System Manager: Administering Avaya Aura Session Manager, Doc ID Avaya Aura Session Manager Overview, Doc ID Maintaining and Troubleshooting Avaya Aura Session Manager, Doc ID of 41

41 Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at 41 of 41