Abstract. Avaya Solution & Interoperability Test Lab

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Esna Officelinx Version 10.6 with Avaya Communication Server 1000 Release 7.6 and Avaya Aura Session Manager Release Issue 1.1 Abstract These Application Notes describe the procedure for configuring Esna Officelinx 10.6 to interoperate with the Avaya Communication Server 1000 Release 7.6 and Avaya Aura Session Manager Release The Officelinx voice server connects to the Avaya Communication Server 1000 through a SIP Trunk connection and provides unified communications features such as the greetings menu, user mailbox services, wake up services and transfer functions. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 1 of 30

2 1. Introduction These Application Notes describe the procedure for configuring Esna Officelinx 10.6 (Officelinx) to interoperate with Avaya Communication Server 1000 R7.6 and Avaya Aura Session Manager R The objective of this compliance testing is to verify that Officelinx connects to the CS1000 via SIP trunks and provides unified communication services such as greetings, messaging and transfer functions. 2. General Test Approach and Test Results The general test approach was to place calls to the Esna Officelinx voice server and verify that the user can: Establish calls between Officelinx and the CS1000E end points. Transfer from Officelinx. Leave and retrieve messages for subscribers. The Esna Officelinx server was tested for serviceability and the objectives were to verify that: Officelinx can successfully re-establish a connection to the CS1000E after the Ethernet cable has been disconnected and reconnected. Officelinx can successfully recover after a reboot Interoperability Compliance Testing The interoperability compliance test included features and services that operate via SIP connectivity. The focus of the compliance testing was primarily to verify the interoperability between Officelinx 10.6 and the CS1000E with regard to the following features: CS1000E end points can access the Officelinx pilot number. Officelinx can access the CS1000E end points. Officelinx provides messaging services to the CS1000E end points. Officelinx can conduct transfer operations for the end points Test Results All tests were performed and passed Support Contact Esna Technical support for further assistance with Officelinx: URL techsupp@esna.com Phone (905) RS; Reviewed: Solution & Interoperability Test Lab Application Notes 2 of 30

3 3. Reference Configuration Figure 1 illustrates the configuration used in these Application Notes. The configuration shows a setup of a CS1000E communicating with Officelinx via Session Manager using SIP trunks. The CS1000E has UNIStim IP and SIP telephones connected as endpoints. For interoperability, Officelinx requires the use of the G.711MU codec, and transmission of DTMF tones using RFC2833. Figure 1: Reference Configuration 4. Equipment and Software Validated The following equipment and software/firmware were used during testing: Equipment Software/Firmware Avaya Communication Server 1000 SW Version : Avaya Telephones: UNIStim IP To be confirmed SIP Avaya Aura Session Manager SW Version : Avaya Aura System Manager SW Version : Officelinx Application Server Windows 2012 R2 or 2016 Officelinx SW Version SU1 RS; Reviewed: Solution & Interoperability Test Lab Application Notes 3 of 30

4 5. Configure Avaya Communication Server 1000 R7.6 This section describes the procedure for setting up the CS1000E. The steps include configuring: Node properties. Routes, Route List Blocks (RLB) and Distant Steering Codes (DSC). Endpoints/Telephones. The values used in this guide are specific to the test environment. Clients must enter values appropriate to their site (e.g. IP addresses, extension numbers, etc.). CS1000E configurations are performed through the Unified Communications Manager (UCM), Element Manager (EM) and Command Line Interface (CLI) via a telnet session to the Call Server Node IP (SIP Gateway) Configuration This section describes the configuration of the SIP Gateway application running on the CS1000E signaling server. In the test environment, Node ID 511 was used since it has the SIP Gateway application enabled. For additional information on Nodes configuration, refer to Section 10. To configure the SIP Gateway from EM, go to System > IP Network > Nodes: Servers, Media Cards and click Node ID 511 as shown in Figure 2. Figure 2: Accessing IP Telephony Nodes RS; Reviewed: Solution & Interoperability Test Lab Application Notes 4 of 30

5 Click Gateway (SIPGw & H323Gw) as shown in Figure 3. Figure 3: Accessing SIP and H323 Gateways On the General tab, enter the SIP domain name as bvwdev.com (from Figure 19), set the Local SIP port to 5060, Gateway endpoint name to cppm1 (from Figure 23a) and Application node ID to 511 (from Figure 2) as shown in Figure 4. Figure 4: Configuration of General Fields RS; Reviewed: Solution & Interoperability Test Lab Application Notes 5 of 30

6 Figure 5 shows the Primary TLAN IP address, which is the IP address of the Session Manager. The other fields can be left at their default values. Figure 5: Configuring the Session Manager IP Address Figure 6 shows the SIP URI Map configuration where the UDP field is configured as udp. The other fields can be left at their default values. Figure 6: Configuring SIP URI Map Fields RS; Reviewed: Solution & Interoperability Test Lab Application Notes 6 of 30

7 5.2. Route, RLB and DSC Configuration This section explains the steps to configure a routing entry that will access the Officelinx server from the CS1000E using the RLB and DSC values. After logging into the UCM, click the EM link of the respective CS1000E (Not Shown). Go to Routes and Trunks > Routes and Trunks. Click Add route as shown in Figure 9. Figure 9: Adding Route Figure 10 shows the configuration of the route being added. The values that are circled in red are configured by the user. The values shown were used during the testing. Figure 10: Route Configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 7 of 30

8 To configure the RLB using EM, go to Dialing and Numbering Plans > Electronic Switched Network > Network Control & Services > Route List Block (RLB) as shown in Figure 11. Figure 11: Accessing RLB Enter the value of the route list index and click the to Add button to continue the configuration as shown in Figure 12. During testing, a value of 10 was used. Figure 12: Adding RLB Figure 13 shows Route Number 10 being selected for the RLB created. Route 10 was selected since it was the route number assigned while adding a route (see Figure 10). Figure 13: Selecting the configured Route to RLB RS; Reviewed: Solution & Interoperability Test Lab Application Notes 8 of 30

9 To configure the DSC using EM, go to Dialing and Numbering Plans > Electronic Switched Network > Coordinated Dialing Plan (CDP) > Distant Steering Code (DSC) (see Figure 11). In the Distant Steering Code List page, select Add from the drop down menu (Figure 14). Figure 14: Adding a new DSC Enter the value of the DSC and click to Add (Not Shown). As shown in Figure 15, a DSC value of 3 was added during testing. The value 3 was configured since the pilot DN of the Officelinx system was Flexible Length number of digits identifies the length of the directory number (DN). During testing, a value of 5 was used. Route List to be accessed for trunk steering code was set to 10 from the dropdown list. This value is selected based upon the RLB created in Figure 12. Figure 15: DSC configuration For additional information on Route, RLB and DSC configuration, refer to Section 10. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 9 of 30

10 5.3. Endpoint/Telephone Configuration This section explains the provisioning of an endpoint/telephone. The Endpoint/Telephone can be configured using the CLI of the CS1000E from overlay LD 11/20. Refer to Section 10 for further information regarding adding and configuring endpoints/telephones. In Figure 16, values that are shown in red are configured by the user. The FDN and HUNT value of was used during testing as the pilot DN of the Officelinx system. Figure 16: Configuring an Endpoint/ Telephone RS; Reviewed: Solution & Interoperability Test Lab Application Notes 10 of 30

11 6. Configure Routing using Avaya Aura System Manager This section provides the procedures for configuring routing using Avaya Aura System Manager. Log onto the Aura System Manager web admin console. The configuration process include the following: Adding a Domain. Adding a Location. Adding SIP entities. Adding Entity links. Adding Routing Policies. Adding Dial Patterns Adding Domain To add a domain, select Domains from the left-hand window of the Routing screen and click New (not shown). Configure the Name as shown in Figure 19 and select Commit to add a domain. During compliance testing, the domain name bvwdev.com was used. Additional domains can be added as required. Figure 19: Domain Management RS; Reviewed: Solution & Interoperability Test Lab Application Notes 11 of 30

12 6.2. Adding Location To add a location, select Locations from the left-hand window of the Routing screen and click New (not shown). Configure the Name as shown in Figure 20 and select Commit to add a Domain. During compliance testing, the location name Belleville, Ont, Ca was used. Additional locations can be added where required. Figure 20: Location Details 6.3. Adding SIP Entities This section explains the adding of SIP entities for the Session Manager, Officelinx and CS1000E system routing. To add SIP Entities, select SIP Entities from the left-hand window of the Routing screen and click New (not shown). Figures 21a and 21b show the SIP Entity Details for the Session Manager routing. The FQDN or IP Address ( ) is the IP address of the Session Manager. Click Commit to add the SIP Entity. Figure 21a: SIP Entity Details for Session Manager RS; Reviewed: Solution & Interoperability Test Lab Application Notes 12 of 30

13 Figure 21b: SIP Entity Details for Session Manager (cont d) Figures 22a and 22b show the SIP Entity Details for Officelinx routing. The FQDN or IP Address ( ) is the IP address of the Officelinx voice server. Click Commit to add the SIP Entity. Figure 22a: SIP Entity Details for the Officelinx System RS; Reviewed: Solution & Interoperability Test Lab Application Notes 13 of 30

14 Figure 22b: SIP Entity Details for the Officelinx System (cont d) Figures 23a and 23b show the SIP Entity Details for the CS1000E System routing. The FQDN or IP Address ( ) is the Node IP address of the SIP Signaling Gateway of the CS1000E System. Click Commit to add the SIP Entity. Figure 23a: SIP Entity Details for CS1000E System Figure 23b: SIP Entity Details for CS1000E System (cont d) RS; Reviewed: Solution & Interoperability Test Lab Application Notes 14 of 30

15 6.4. Adding Entity Links This section explains the addition of Entity links between the Session Manager and the CS1000E. To add an entity link, select Entity Links from the left-hand window of the Routing screen and click New (not shown). Figure 24 shows the Entity Link added between the Session Manager and CS1000E. Figure 24: Adding Entity Link An Entity Link is also added between the Session Manager and the Officelinx voice server Adding Routing Policies This section explains the Routing Policy configuration for Officelinx and CS1000E Systems. To add a routing policy, select Routing Policies from the left-hand window of the Routing screen and click New (not shown). Figures 25a and 25b show the Routing Policy Details for Officelinx. Select the Officelinx System as the SIP Entity Destination and add the dial pattern associated with it. A dial pattern can be added once it has been configured (see Section 6.7). Click Commit to add a routing policy. Figure 25a: Routing Policy Details for Officelinx System RS; Reviewed: Solution & Interoperability Test Lab Application Notes 15 of 30

16 Figure 25b: Routing Policy Details for Officelinx System (cont d) Above steps can be repeated with the required fields to configure routing policies for the CS1000E. During compliance testing, a dial pattern of 58 was configured for the CS1000E since the extension series on the CS1000E was in the 58xxx range. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 16 of 30

17 6.6. Adding Dial Patterns This section explains the steps to add dial patterns to the Officelinx and CS1000E systems. To add a dial pattern, select Dial from the left-hand window of the Routing screen and Dial Patterns. Click New (not shown). Figure 26 shows the Dial Pattern Details for Officelinx. During compliance testing, the extension range on Officelinx began with 3xxxx. Therefore, 300 was used in the Pattern field. The minimum and maximum size of the extension is set to 5. Add the ESNA_Routing policy (configured in Section 6.6). Click Commit to add the dial pattern. Additional dial patterns can be configured as required. Figure 26: Dial Pattern Details 7. Configuring the Esna Officelinx server The Officelinx installation is covered in referenced product documentation. During the install, the PBX template Avaya is selected providing a pre-defined configuration. This section only describes the interface configuration as a reference for verification, so that Officelinx can communicate with the CS1000E. For further details on Officelinx configuration not covered here, refer to Section 10. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 17 of 30

18 Select Avaya as the PBX type. Select Communication Server 1000 SM as the PBX model. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 18 of 30

19 Select None if no server is configured. Enter the required information to go through AES server configuration. The Nodes button becomes active once the required information has been entered. Dummy information can be used if no AES is configured. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 19 of 30

20 Click Next. Configuring Officelinx to integrate with CS1000E is done through the Officelinx SIP Configuration Tool. To access the SIP Configuration Tool on the Officelinx voice server, go to Start > All Programs > Officelinx > SIP Configurator (Not shown. The exact steps will vary based upon your Windows version). RS; Reviewed: Solution & Interoperability Test Lab Application Notes 20 of 30

21 Figure 27 shows the SIP Configuration tool when a new PBX is added. During testing, the template Nortel was selected as the PBX. Figure 27: Adding new PBX RS; Reviewed: Solution & Interoperability Test Lab Application Notes 21 of 30

22 Figure 28 shows the General tab of the SIP Configuration Tool. Fields circled in red are entered by the user. The IP Address field is the IP address of the Session Manager. Figure 28: General Configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 22 of 30

23 Figure 29 shows the Advanced tab of the SIP Configuration Tool. Enable the checkboxes as shown. Figure 29: Advanced Configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 23 of 30

24 Figure 30 shows the pilot DN of the Officelinx voice server configured in the Channels tab of the SIP Configuration Tool. During testing, was configured as the Officelinx pilot DN. Figure 30: Channels Configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 24 of 30

25 Figure 31 shows the Force MWI box enabled under the MWI tab. The remaining values can be left at their defaults. Figure 31: MWI Configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 25 of 30

26 The integration mode number for Officelinx to communicate with CS1000E is 4. This Integration Mode field can be configured from the ANI tab of the SIP Configuration Tool (see Figure 32). Click OK to complete the configuration. Figure 32: ANI Configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 26 of 30

27 Figure 33 shows the General Settings tab. Due to the large packet size of SIP messages from the Session Manager, the Buffer Size must be set as This changes the default TCP packet size from 2kb to 4kb. Figure 33: General setting configuration RS; Reviewed: Solution & Interoperability Test Lab Application Notes 27 of 30

28 Officelinx must be configured for SIP to send the Message Waiting Light indication to the PBX. Launch OL Admin through Start > All Programs > Officelinx > Office Linx Admin (Not Shown). Access the Properties of the PBX as shown in Figure 34. Figure 34: Accessing PBX Properties Screen RS; Reviewed: Solution & Interoperability Test Lab Application Notes 28 of 30

29 Enable the SIP radio button from the Message Light tab shown in Figure 35. Figure 35: Message Light Configuration Add TCP 5060 to the firewall as an allowed port into the Officelinx. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 29 of 30

30 8. Verification Steps The following steps may be used to verify the integration: From the CS1000E end point, call the Officelinx pilot DN and verify that the general greeting is played. Verify that a call from a CS1000E endpoint to another CS1000E endpoint can be transferred via Officelinx server. Test that the correct Officelinx greeting messages are played depending upon the status of the CS1000E endpoints. Check that messages can be left for a CS1000E endpoint and retrieved through the Officelinx server. 9. Conclusion All of the test have passed, meeting the objectives outlined in Section 2. The Esna Officelinx 10.6 software is considered compliant with Avaya Communication Server 1000E Release7.6 and Avaya Aura Session Manager Release Additional References Additional Avaya product documentation is available at [1] Software Input Output Reference Administration Avaya Communication Sever 1000, R7.5 NN , Sept 2011 Product documentation for Esna Officelinx may be found at: Avaya Inc. All Rights Reserved. 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 DevConnect Program at devconnect@avaya.com. RS; Reviewed: Solution & Interoperability Test Lab Application Notes 30 of 30