Lotusphere IBM Collaboration Solutions Development Lab

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1 Lotusphere 2012 IBM Collaboration Solutions Development Lab Lab#4 IBM Sametime Unified Telephony Lite telephony integration and integrated telephony presence with PBX 1

2 Introduction: IBM Sametime Unified Telephony Lite Client (SUT Lite) provides a new SIP trunking feature for integrating Sametime with third-party audio and video equipment, which enables Sametime Connect client to make and receive point-to-point calls. Deployment of SUT Lite is an excellent entry point into unified communications (UC) in your organization. Sametime also has a variety of extension toolkits that can be used to add new UC features on Sametime. Integrated telephony presence can be added using IBM Sametime Connect client toolkit and Community server toolkit technology to show telephony status along with Sametime presence. This lab provides SUT Lite and Sametime Toolkit development environments. You will learn how to: 1) Configure Sametime server to provide a connectivity of the Sametime to the external telephony system. 2) Extend Sametime server and client with Sametime toolkit to show integrated presence capability. Objective: In this lab you will learn how to configure and enable point-to-point call integration of Sametime with third-party telephony equipment using the SUT Lite. In addition you can also learn how to extend Sametime Connect client using the extension toolkit to integrate telephony presence. This lab is broken out into 2 Exercises. Exercise 1 covers configuration of SUT Lite environment for trunking with Asterisk, and Exercise 2 cover extending Sametime Connect client presence feature with adding Integrated telephony presence. Note: This lab environment is consist from two server machines (Sametime server and Asterisk server) and one client machine. The Sametime server machine contains the following Sametime server components: IBM Sametime Community server IBM Sametime Community server is installed on top of IBM Lotus Domino server which is installed to C:\Lotus\Domino folder. IBM Sametime System Console Server IBM Sametime Media Manager IBM Sametime System Console server and Media Manager are installed on top of IBM WebSphere Application server which was installed to C:\IBM\WebSphere folder. The Asterisk server machine contains the following Asterisk distribution component: trixbox CE

3 The client machine contains following software IBM Sametime Connect client IBM Sametime Connect client is installed to C:\Lotus\Sametime Connect folder. Lotus Domino Administrator and Lotus Notes Lotus Domino Administrator and Lotus Notes are installed to C:\Lotus\Notes folder. X-Lite 4.1 Eclipse IDE for Java Developers, Indigo Service Release 1 Eclipse IDE binary is extracted to C:\eclipse folder. vmsutlite.lotuslabs.com IBM Sametime server IBM Sametime System Console IBM Sametime Community IBM Sametime Media Manager vmasterisk.lotuslabs.com Asterisk server (trixbox CE) IBM Sametime Connect client X-Lite (Softphone for Asterisk) Eclipse IDE Figure 1: Lab machines configuration 3

4 Exercise 1: Integrating SUT Lite with Asterisk Background information: IBM Sametime introduces IBM Sametime Unified Telephony Lite Client (SUT Lite), which provides a new SIP trunking feature for the integration of the Sametime and third-party audio and video equipment used within your organization. SUT Lite is a separately licensed feature that builds on your Sametime Standard deployment to add SIP-based telephony to the feature set. SUT Lite's SIP trunking feature provides the ability for a Sametime Connect client to make and receive point-to-point calls with external SIP endpoints such as Asterisk PBX. When the Sametime Connect user dials a phone number, SUT Lite establishes a SIP-based telephone call, and the user can talk over the softphone embedded in Sametime Connect client. A SIP trunk is a direct IP network connection between the Sametime Media Manager's SIP Proxy/ Registrar component and an external PBX's SIP endpoint. During deployment of SUT Lite, the Sametime Administrator defines a unique static route that the SIP trunk uses for telephony connection. Setup Information: Before you begin configuring SUT Lite, you should figure out how calls from the Sametime will connect to the third-party SIP endpoint by working out a dial plan. A dial plan is the set of dial strings (phone numbers), rules (conditions), and SIP endpoints (destinations) that enable one user to place a telephone call to another user. The following table shows a dial string pattern, the name of the SIP endpoint that services calls matching that pattern, and the name of the end-user device that receives the call. This will help you define routes and routing rules. The dial string specifies a pattern that the user must exactly match when initiating a call to a particular device. In the dial string, each "X" represents exactly one digit from 0 through 9. For example, if you use the mapping shown in the table, 1XXX means that the Sametime user could dial four-digit number between 1000 to 1999 and this four-digit number would route to the Asterisk, and from there it would be directed to a user's desk phone. Dial String SIP endpoint Phone/Device 5XXX Sametime SIP Proxy/Registrar Sametime softphone 1XXX Asterisk Asterisk desk phones Table 1: Dial plan in lab environment SUT Lite is a separate licensed application that adds SIP trunking functionality to a Sametime standard deployment. You must purchase and activate the SUT Lite license, then you have to enable SUT Lite feature in a policy setting on your Sametime deployment. In the lab environment, all of required software were already installed and enabled. And configurations are also done. You can verify the SUT and Asterisk configuration during this exercise. 4

5 Hint: Learn more details about the IBM Sametime Unified Telephony Lite Client from the following link. lookupname=administering%20sametime%20unified%20telephony%20lite %20Client% %20documentation 5

6 Procedures: Enabling Sametime Unified Telephony Lite Client Before connecting your Sametime with an external PBX system, you have to enable SUT Lite features on the Sametime server. In the lab environment, All of required components for SUT Lite were already installed and all of the enablement procedures described in this section were already completed. 1. Enabling the SUT Lite feature Enabling the SUT Lite feature allows the Sametime Media Manager to support the SIP trunking feature implemented in SUT Lite, so that Sametime clients can make calls to third-party SIP endpoints, which in turn routes calls to end-user devices such as desk phone, softphones, and conference room video systems. Note: The enablement procedure described in this step was already completed in this lab environment. You can follow the procedure and ensure the setting. a) From a browser, open WebSphere System Console login page with the following URL: Figure 2: WebSphere System Console login page b) Log in to the Integrated Solutions Console using the following WebSphere Application server administrator ID and password: Username: wasadmin Password: l0tus2012 c) On the navigation tree, click Sametime System Console > Sametime Servers > Sametime Media Manager. d) click the stmedia in Deployment Identifier list of the Sametime Media Manager. 6

7 Figure 3: Deployment Identifier list of the Sametime Media Manager e) On the Configuration page, locate the Sametime Unified Telephony Lite section, select Allow the use of Sametime Unified Telephony Lite clients, and then click OK button. Figure 4: Selecting Allow the use of Sametime Unified Telephony Lite clients 2. Modifying the Sametime policy to allow SIP trunking calls As the SUT Lite feature has been enabled, we will set the policy that allows clients to use SIP trunking so user can take advantage of the new feature. Note: The enablement procedure described in this step was already completed in this lab environment. You can follow the procedure and ensure the setting. a) On the navigator, click Sametime System Console > Manage Policies. b) On the Policies page, click the Media Manager tab. c) On the Media Manager tab, select Sametime Media Manager Default Policy in the table, then click the Edit button at the top of the table. 7

8 Figure 5: Selecting Sametime Media Manager Default Policy and edit it d) On the Media Manager default policies page, locate the Sametime Unified Telephony Lite Client section, select Allow calls that use SIP Trunk capability, and then click OK button. Figure 6: Selecting Allow calls that use SIP Trunk capability 3. Verifying the port used by the Sametime Media Manager a) From a browser, log in to the Integrated Solutions Console as the WebSphere administrator (it may already be open from the last task). b) In the navigation tree, click Servers > Server Types > WebSphere application servers. c) In the WebSphere application servers list, click STMediaServer link to open configuration page. 8

9 Figure 7: Clicking STMediaServer in WebSphere Application servers list d) On the Configuration page, locate the Communications section and click Ports link. Figure 8: Ports link in the Communications section e) In the Ports table, look for the port used by the Conference Manager, and write down the value. in this lab we are configuring TCP (unsecured) communications: SIP_DEFAULTHOST Figure 9: Ports used by the Conference Manager f) Next, check the port used by the SIP Proxy/Registrar, and write the value down. In this lab, we are configuring TCP (unsecured) communications: SIP_ProxyRegHOST 9

10 Figure 10: Ports used by the SIP Proxy/Registrar Note: In the lab environment, secure connections of SIP trunk were disabled for the exercise purpose. Please use insecure values for the communication port numbers. Configuring the telephone number attribute in the LDAP 4. Verify the assigned telephone numbers for users a) Launch Lotus Domino Administrator and login with the following ID Username: admin/lotuslabs Password: l0tus2012 b) Click LOTUSLABS Domain tab and select Domino Directories > LotusLabs's Directory > People in the navigation tree c) Observe that user5001, user5002 and user5003 have assigned office phone numbers: 5001, 5002 and 5003 for each. Figure 11: People list in Domino Directory 10

11 d) Select File > Exit Administrator to terminate the Domino Administrator. Configuring the inbound and outbound routes for Sametime 5. Configuring the inbound route to Sametime The SIP Proxy/Registrar uses inbound routing rules to receive call requests from third-party audio/video devices and then direct those calls to Sametime clients. In the lab environment, you will set up a static inbound route with a routing rule that tells the SIP Proxy/Registrar to direct all calls from the Asterisk's IP address to the Sametime Media Manager's Conference Manager component. Note: The inbound routes to Sametime are already created on the lab environment. You can just see the existing routing rule. a) From a browser, log in to the Integrated Solutions Console as the WebSphere administrator. b) In the navigation tree, click Sametime System Console > Sametime Servers > SIP Proxies and Registrars. c) On the SIP main menu page, click the stmedia link in Deployment Identifier list of the SIP Proxy and Registrar. Figure 12: Deployment Identifier list in SIP Proxy and Registrar d) On the stmedia SIP Proxy and Registrar page, click Proxy Administration link. Figure 13: Administration options in stmedia SIP Proxy/Registrar 11

12 e) In the routing rules table, click New button to create a new rule. Figure 14: Empty Routing Rules list f) On the Add New Rule page, type Inbound Asterisk as a name of the new inbound route in the Name field. g) In the Conditions section, configure the routing rule as the following: Method: INVITE Source Address: checked IP address of Asterisk server in you lab environment. 12

13 Figure 15: New routing rule for inbound routing from Asterisk Note: The IP address for Asterisk server may change depending on the lab environment set you are using. You can find the IP address for Asterisk server by the following command in Command Prompt Window: > ping vmasterisk.lotuslabs.com h) In the Destination section, configure the destination as the following: Select Push a Route header field radio button. Using fields: checked Scheme: SIP IP/FQDN: vmsutlite.lotuslabs.com Port: 5063 Transport: TCP 13

14 Figure 16: Destination section in new Inbound Asterisk routing rule i) Click OK button to save the new inbound routing rule. 6. Configuring the outbound route to Asterisk The SIP Proxy/Registrar uses outbound routing rules to receive call requests from Sametime clients and then direct those calls to third-party SIP endpoints, which in turn routes calls to the callee s audio/video devices. For the example configuration, you will set up a static outbound route with a routing rule that tells the SIP Proxy/Registrar to direct all outgoing calls to Asterisk. Configure an outbound route from Sametime to Asterisk as follows: Note: The outbound routes for Asterisk are already created on the lab environment. You can just see the existing routing rule. a) From a browser, log in to the Integrated Solutions Console as the WebSphere administrator. b) In the navigation tree, click Sametime System Console > Sametime Servers > SIP Proxies and Registrars. c) On the SIP main menu page, click stmedia link in Deployment Identifier list of the SIP Proxy/Registrar. d) On the SIP Proxy and Registrar page, click Proxy Administration link. e) In the routing rules table, click New button to create a new rule. f) On the Add New Rule page, type Outbound Asterisk as a name of the new outbound route in the Name field. g) In the Conditions section, configure the routing rule as the following: 14

15 Method: Request URI: INVITE checked sip:1.* The string specified in the Request URI field is a regular express which resolves the SIP URL for endpoint on the Asterisk server. As.* matches any number of characters, sip:1.* matches any string which starting with sip:1. Therefore, the request URL specified here would match the SIP URI for Asterisk desk phones like sip:1001@vmasterisk.lotuslabs.com. Figure 17: New routing rule for Outbound routing to the Asterisk server h) In the Destination section, configure the routing rule as the following: Select Replace a Request-URI radio button. Using a regular expression: checked Request-URI pattern: sip:(.+)@.* Outbound pattern: sip:$1@vmasterisk.lotuslabs.com To achieve outbound route, the request URI must be changed to a format that Asterisk will accept. The SIP requests for outgoing calls are generated by the Sametime client using the format sip:1xxx@vmsutlite.lotuslabs.com where "1XXX" is the telephone number to call. If the request is routed to Asterisk as-is, it will result in a 404-User not 15

16 found error because the Asterisk can only process a requests that use its own domain and will reject all other requests. To resolve this, your route definition will modify the request URI by replacing the Sametime domain with the Asterisk domain. This is accomplished by the Request URI pattern configuration. In the Request URI pattern, the expression within the parentheses (.+) will be assigned to a corresponding variable $1 in the Output pattern. In the case of this regular expression specified in Request-URI pattern, everything between "sip:" and "@" will be assigned to variable $1. The Output pattern will copy that information into a new URI format that uses the Asterisk domain before sending the modified request to the Asterisk server. For example, if the Sametime Request URI looks like this: sip:1001@ vmsutlite.lotuslabs.com then it will be modified to look like this: sip:1001@vmasterisk.lotuslabs.com Figure 18: Destination section in new Outbound Asterisk routing rule i) Click OK button to save the new outbound route. Now as you have configured the inbound and outbound routing rules for Sametime, restart the SIP Proxy/Registrar server. j) In the server s Integrated Solutions Console, click Servers > Server Types > WebSphere application server. k) In the WebSphere application server list, select the check box for STMediaServer and click the Restart button at the top of the table. 16

17 Figure 19: Restarting STMediaServer in the WebSphere application server list l) Wait until all servers' status will change to Started. Configuring Asterisk 7. Verifying the current extension settings trixbox is an Asterisk distribution with including convenience Web base administrative user interface. In this lab, trixbox is configured as sample of existing corporate's PBX, and all extensions which will be used in this lab are already registered. The registered extension can be verified by the following procedures: a) From a browser, open the following URL to access Asterisk (tirxbox) administration page. b) Login the Asterisk administration page with following user name and password. Username: maint password: password c) Select PBX > PBX Settings at menu bar and click Setup > Basic > Extensions. 17

18 Figure 20: Opening PBX Settings in Asterisk administration page d) As the existing extensions are listed at right side, You can click TestUser1001<1001> to see its detail setting. Figure 21: Extension management in Asterisk administration page 8. Configuring SIP Trunk to Sametime Note: The SIP Trunk is already created in the lab environment. You can just see the existing SIP Trunk. 18

19 a) On the Asterisk administration page, select PBX > PBX Settings at menu bar and click Setup > Basic > Trunks. b) Click Add SIP Trunk link. Figure 22: Adding new SIP Trunk in Asterisk administration page Note: The existing SIP Trunks are listed in right side of this page. You can click Trunk SIP/SUT Lite to see the current setting. The following steps show how to create new SIP Trunk, but you can skip them in the lab environment. c) On Add SIP Trunk page, specify the following: Outbound Caller ID: Asterisk Dial Rules: <blank> Outgoing Setting section Trunk Name: SUTLite PEER Details: Incoming Settings section USER Context: USER Details: host=vmsutlite.lotuslabs.com port=5080 type=peer allow=all allowtransfer=yes User type=user context=from-trunk 19

20 Figure 23: Specifying the trunk name in Add SIP Trunk page Figure 24: Outgoing Setting section in Add SIP Trunk page Figure 25: Incoming Settings in Add SIP Trunk page d) Click Submit Changes button. e) Click Apply Configuration Changes button at the top of the administration page, and click Continue with reload link. 20

21 Figure 26: Clicking Apply Configuration Changes button Figure 27: Clicking Continue with reload link to apply changes 9. Configuring the Outbound Route to Sametime Note: The Outbound Route to Sametime is already created in the lab environment. You can just see the existing Outbound Route. a) On the asterisk administration page, select PBX > PBX Settings at menu bar and click Setup > Basic > Outbound Routes Note: The existing Routes are listed in right side of this page. You can click 1 SUTLite to see the current setting. The following steps show how to create new Outbound Route, but you can skip them in the lab environment. b) On Add Route page, specify the following: Route Name: SUTLite Dial Patterns: 5XXX Trunk Sequence: SIP/SUTLite 21

22 Figure 28: Adding new Outbound Route for Sametime server c) Click Submit Changes button 10. Configuring Asterisk to allow anonymous inbound SIP calls Note: This configuration is already done in the lab environment. You can just see the current configuration. a) On the Asterisk administration page, select PBX > PBX Settings at menu bar and click Setup > Basic > General Setting. b) In General Setting section of the page, select yes for Allow Anonymous Inbound SIP Calls?. Figure 29: Enabling Allow Anonymous Inbound SIP Calls? c) Click Submit Changes button. d) Click Apply Configuration Changes button at the top of the administration page, 22

23 and click Continue with reload link. 23

24 Verifying the interoperability between Sametime and Asterisk Now all of configuration on Sametime and Asterisk were completed. You are ready to test the interoperability between the Sametime softphone and telephone device under the Asterisk server. We will use X-Lite softphone as a telephone device in the Asterisk domain. 11. Verifying the interoperability with Sametime Connect and softphone a) Double click X-Lite icon on the desktop to launch X-Lite softphone client. Figure 30: X-Lite icon b) If X-Lite is not connected to Asterisk automatically with existing account setting, select Softphone > Account Setting to open SIP Account dialog box and specify the following account setting, then click OK button: Account name: Asterisk User Details section User Id: 1001 Domain: lotuslabs.com Password: l0tus2012 Domain Proxy section select Proxy Address: vmasterisk.lotuslabs.com c) Double click Sametime Sametime Connect client icon on the desktop to launch Lotus Sametime Connect client. Figure 31: Sametime Connect icon a) On Sametime Connect client, login to Sametime with the following login information. Host server: vmsutlite.lotuslabs.com User name: user5001 Password: l0tus2012 b) On Sametime Connect client, click Call options > Call Phone Number. 24

25 Figure 32: Selecting Call Phone Number... c) Type 1001 in Enter Phone Number dialog and click Call Number button. Figure 33: Sametime dialler Figure 34: Sametime Dialog shows call status d) As X-Lte client receives an incoming call, click Audio button to answer the call, then click End button to terminate the call. 25

26 Figure 35: X-Lite notices an incoming call and accept it by clicking Audio button e) On X-Lite client, type 5001 in the call number field and click Call button. Figure 36: Making a call to Sametime user from X-Lite f) As Sametime Connect client receives an incoming call, click Answer button to answer the call, then click Leave Call button to terminate the call. Summary: In this lab, you learned how to configure Sametime server and Asterisk server to communicate via SIP trunking. On both server, static routes for SIP trunk are defined and inbound and outbound routings were configured. Although you have to create detailed numbering plan to deploy SUT Lite environment into your organization's real telephony environment, the routing configuration on Sametime are very simple and SUT Lite can be easily integrated into the exiting telephony network in your organization. 26

27 Exercise 2: Build your Integrated Telephony Presence Background information: IBM Sametime Connect client includes the telephony status plug-in, which uses the capabilities of the livename's extended status API to display telephony status icons on livenames in the contact list and elsewhere. The telephony status plug-in works in conjunction with the Lotus Sametime Telephony Presence Adapter in Sametime Community Server Toolkit. The Telephony Presence Adapter can be used by third-party telephony service providers to show customized telephony presence. The Lotus Sametime server obtains telephony status from the Telephony Presence Adapter and publishes status changes via user attributes, which will be displayed on Sametime Connect client. In this lab, we will use Asterisk Manager API to monitor the telephony channels for Sametime spoftphone and detect telephony usage status of the Sametime client. With using this information, we will extend the Telephony Presence Adaptor program to show real telephony presence on Sametime Connect client. Setup Information: In this lab, you will try the sample program for Telephony Presence Adapter at first. The sample program and required Java library files can be found in IBM Sametime Toolkit (Software Development kit) at C:\LabFiles\st852sdk. Asterisk Manager API allows an external software to connect to an Asterisk over TCP/IP sockets on port 5038 and send commands and receive events from Asterisk server. We will modify the Telephony Presence Adaptor sample program and integrate it with Asterisk Manager API to detect telephony event on the Asterisk server. In this lab, we will use our original simple implementation of Java wrapper to communicate with Asterisk Manager API port. The Asterisk Manager Java wrapper can be found at C:\LabFiles\java\AsteriskManagerAPI.zip. Hint: Learn more details about the IBM Sametime telephony status plug-in and Telephony Presence Adapter from the following documentarians: Sametime Connect Toolkit Integration Guide (st852sdk\client\connect\doc\st_integration_guide.pdf in the toolkit folder) Sametime Community Server Toolkit Developer s Guide (st852sdk\server\commserver\doc\stcommsrvrtkdevguide.pdf in the toolkit folder) IBM Sametime Software Development kit are found at the following link: Hint: Learn more details about the Asterisk Manager API from the following link. 27

28 Setup development environment and try Telephony Presence Adaptor sample program 1. Enabling Sametime Connect for Telephony Status plug-in a) Open the Sametime plug-in configuration file C:\Lotus\Sametime Connect\rcp\plugin_customization.ini with NotesPad and verify that the following line was inserted. This property enables Sametime Connect client to show the Telephony Presence. com.ibm.collaboration.realtime/enabletelephonystatus=true Note: Although the setting described in this step is not enabled by default configuration, it was already completed in the lab environment. Detail for enabling the telephony presence on Sametime Connect client is described in the following Sametime Wiki article: extended_status_and_telephony_status_in_the_client_st852 b) Close the plugin_customization.ini file c) Then double click Sametime Connect client icon on the desktop to launch Lotus Sametime Connect client. Figure 37: Sametime Connect icon 2. Launch Eclipse Integrated Development Environment (IDE) a) Double click Eclipse IDE icon on the desktop to launch Eclipse IDE. Figure 38: Eclipse icon b) In the Workspace Launcher panel, enter workspace path as C:\work\XXworkspace where XX is your initial, and click OK button. 28

29 Figure 39: Selecting workspace for Eclipse IDE c) If you see a Welcome page, close it by clicking [x] button. Figure 40: Closing the welcome page 3. Create a project for building the Telephony Presence Adaptor sample program a) From Eclipse IDE menu, select Files > New > Java project. b) On Create a Java Project dialog, specify the project name as Telephony Adaptor, and click Finish button. 29

30 Figure 41: Create a Java Project dialog box c) Locate the TelephonySample.java file in C:\LabFiles\st852sdk\server\commserver\samples\telephony folder in File explorer, and drag & drop the file into the src folder in Telephony Adaptor project in Eclipse IDE. If File Operation dialog is displayed, select Copy files and click OK button. Figure 42: File Operation dialog 30

31 d) Locate stcommsrvrtk.jar file in C:\work\Sametime\st852sdk\server\commserver\bin folder in File explorer, and drag & drop the file into the root folder in Telephony Adaptor project in Eclipse IDE. If File Operation dialog is displayed, select Copy files and click OK button. Now your Eclipse IDE environment looks like the following: At this point, you would see some errors in your Java project. These errors will be fixed in the following procedures. Figure 43: Telephony Adaptor project with compile error e) On Eclipse IDE, select the Telephony Adaptor project and click right mouse button to show a context menu. Select Properties in the context menu to show Properties for Telephony Adaptor dialog. f) On the Properties for Telephony Adaptor dialog, select Java Build Path in left navigator and Library tab in right panel. Then Click Add Jars button to show JAR selection dialog. Figure 44: Properties for Telephony Adaptor dialog, Java Build Path page 31

32 g) Select stcommsrvrtk.jar and click OK on Jar Selection dialog box. Figure 45: Selecting stcommsrvrtk.jar in JAR Selection dialog box h) On the Properties for Telephony Adaptor dialog, click OK on the Properties for Telephony Adaptor dialog. Figure 46: Properties for Telephony Adaptor dialog, Java Build Path You will return the Telephony Adaptor project and no error should be shown. 32

33 Figure 47: Telephony Adaptor project with compile no error 4. Configure the setting a) Locate the st.telephony.adapter.properties file in C:\LabFiles\st852sdk\server\commserver\samples\telephony folder in File explorer, and drag & drop the file into the src folder in Telephony Adaptor project in Eclipse IDE. If File Operation dialog is displayed, select Copy files and click OK. Figure 48: Telephony Adaptor project with properties file b) On Eclipse IDE, double-click st.telephony.adapter.properties file to open it and modify the connecting.server.dns property value as the following: connecting.server.dns=vmsutlite.lotuslabs.com 33

34 c) Select File > Save from menu to save the properties file, then select File > Close to close it. 5. Specify the target user on whom the telephony presence will be displayed. In the telephony sample application, the target user, on whom telephony status will be displayed, is hard-coded as sample name. You have to change the name into the existing user name in your Sametime server. a) Open TelephonySample.java file in src > (default package) folder and locate the definition of USER_TELEPHONY_ID variable. b) Change the value for the USER_TELEPHONY_ID to user5001 as the following: /** * The user on which we will add watch on and publish its telephony status */ private static final String USER_TELEPHONY_ID = "user5001"; c) Select File > Save from menu to save the Java program, then select File > Close to close it. 6. Configure the Sametime server to accept a server application login As the TelephonySample.java program access to the Sametime server as a server application from your machine, you have to add your machine's IP address to the server trusted IP list. f) From a browser, enter the following URL to open WebSphere System Console login page: g) Log in to the Integrated Solutions Console using the WebSphere Application Server administrator ID and password: Username: wasadmin Password: l0tus2012 h) On the navigation tree, click Sametime System Console > Sametime Servers > Sametime Community Server. i) Click stcomm link in Deployment Identifier list of the Sametime Community Servers. 34

35 Figure 49: Deployment Identifier list of Sametime Community Servers j) Locate the New IP address input filed in Trusted Servers session. You can add your machine's IP address here. In the lab environment, your machine's IP is already added and you can just see it. Figure 50: Verifying that your machine's IP is registered in Trusted Servers section 7. Execute the sample program and verify its behavior a) Select TelephonySample.java file in Telephony Adaptor project on Eclipse IDE and click right mouse button to show context menu. Select Run As > Java Application to execute the sample program as a Java application. Note: As a Run Configuration is created with this operation, you can start the same program from the Run History list next time. b) When the program has started and published the telephony status to the Sametime server, a telephony busy icon is displayed at the next of standard Sametime presence for user

36 Figure 51: Sametime Connect client shows the telephony presence 8. Click terminate icon in Console view of Eclipse IDE to stop the program. Figure 52: Terminate icon in Console view on Eclipse IDE Try Asterisk Manager API Asterisk telephony system provides Asterisk Manager API, which provides a method for external software to communicate Asterisk server and add new functionality on Asterisk. Manager API is used to control functionality of the Asterisk system from remote external software over a TCS/IP socket to monitor call activity and perform basic call controls such as initiating and hanging up calls. The Asterisk Manager API allows an external software to connect to an Asterisk over TCP/IP sockets on TCSP port 5038, and send commands and receive events from Asterisk server. There are several implementation which wrap Asterisk Manager API with several programming languages like Asterisk-Java ( ). However we will use our original simple implementation of Java wrapper, which can be found at C:\LabFiles\java\AsteriskManagerAPI.zip. 9. Import Asterisk API sample program and try it a) On Eclipse IDE, select File > Import from menu to open Import dialog. b) On Select dialog box, select Existing Projects into Workspace and click Next button. 36

37 Figure 53: Selecting Existing Projects into Workspace in Import dialog box c) On Import Projects dialog, select Select archive file and select AsteriskManagerAPI.zip at C:\LabFiles\project folder. 37

38 Figure 54: Import project dialog box d) Make sure that Asterisk Manager API project is selected, click Finish button. Now Asterisk Manager API project has been created in you Eclipse workspace. 10. Configure the setting a) Double-click asterisk.managerapi.properties file in Asterisk Manager API project on Eclipse IDE to open it and observe the current property values as the following: server=vmsutlite.lotuslabs.com username=admin secret=amp111 Note: The properties specified in this properties file are used for sample program to login Asterisk Manager API Interface. The authorized username and password for the Asterisk Manager API are configured by manager.config in /etc/asterisk 38

39 directory of the Asterisk server. b) Select File > Save from menu to save the properties file, then select File > Close to close it. 11. Execute the sample program and verify its behavior a) Select AsteriskManagerAPISample.java file in src > (default package) folder of Asterisk Manager API project on Eclipse IDE and click right mouse button to show context menu. Select Run As > Java Application to execute the sample program as a Java application. b) When the program has started and connected to the Asterisk server, the sample program gets the status changes of telephony extensions on the Asterisk server and prints the status information. If you makes a call from Sametime Connect client to the Asterisk phone, the program prints the following informational massages. Nov 16, :04:49 AM com.ibm.ysl.lotus.asterisk.managerapi.managerconnection handleresponse INFO: response> Response = Success Message = Authentication accepted Asterisk Call Manager/1.1 Extension 1001 is ringing Extension 5001 is linked Extension 1001 is linked Extension 1001 is in use Extension 5001 is unlinked Extension 1001 is unlinked Extension 1001 is idle Extension 1001 is ringing Extension 1001 is in use Extension 5001 is linked Extension 1001 is linked Extension 5001 is unlinked Extension 1001 is unlinked Extension 1001 is idle Note: As Sametime softphones are not directly connected to Asterisk server, Asterisk Manager API can returns the status changes of these softphones' extension as linking status of SIP trunk. c) Click terminate icon to stop the program. Integrating Telephony Presence Adaptor with Asterisk Manager API We will use Asterisk Manager API to monitor the telephony channel for Sametime and detect telephony usage status of the Sametime client. With using the telephony status information from the Asterisk Manager API, we will extend the Telephony Presence Adaptor program to show real telephony presence on Sametime Connect client. 39

40 12. Add Java code to make a integration with the Asterisk Manager API sample a) Locate TelephonyStatusController.java in C:\LabFiles\code folder in File explorer, and drag & drop the file into the src folder in Telephony Adaptor project in Eclipse IDE. If File Operation dialog is displayed, select Copy files and click OK button. Figure 55: Additional Java file was copied in to the project b) On Eclipse IDE, select the Telephony Adaptor project and click right mouse button to show a context menu. Select Properties in the context menu to show Properties for Telephony Adaptor dialog. c) On the Properties for Telephony Adaptor dialog, select Java Build Path in left navigator and Projects tab in right panel. Then Click Add button to show Required Project Selection dialog. 40

41 Figure 56: Adding project to Java Build Path d) Select Asterisk Manager API project in the Required Project Selection dialog and click OK. Figure 57: Required Project Selection dialog box e) As you have returned to Properties for Telephony Adaptor dialog, click OK button. 41

42 Figure 58: Asterisk Manager API project has been added. f) You have returned to the Telephony Adaptor project and verify that there is no error in the project. 42

43 Figure 59: Telephony Adaptor project with no error 13. Implement the integration of Telephony Presence Adaptor a) Open TelephonySample.java file in src > (default package) folder of Telephony Adaptor project and locate the definition of publishtelephonystatus() method. b) Change the contents of the publishtelephonystatus() method as the following: /** * Publish a telephony status of the user */ private void publishtelephonystatus() { TelephonyStatusController contoller = new TelephonyStatusController(_service); contoller.exec(); } Note: The program code for this method can be found in TelephonySample_publishTelephonyStatus.java.txt file located at C:\LabFiles\code folder. 43

44 c) Select File > Save from menu to save the Java program file, then select File > Close to close it. d) Locate asterisk.manager.api.properties file in Asterisk Manager API project and copy it to the root of Telephony Adaptor project. 14. Execute the modified Telephony Adaptor program and verify that Sametime Connect client shows the telephony presence. a) Select Run > Run History > TelephonySample from menu to execute TelephonySample.java program. Figure 60: Executing TelephonySample.java from Run History menu Note: You can also run TelephonySample.java Java program by the pull-down menu from Run History action button. Figure 61: Executing TelephonySample.java from Run History icon b) When the program has started and connected to the Asterisk server, the sample program gets the status changes of telephony extensions on the Asterisk server and publishes the updated telephony status to Sametime server. The telephony status will be shown as Telephony Presence on Sametime Connect client. If you have made a call from Sametime Connect client or received a call to, the telephony presence would be changed according to the telephony status. 15. Click terminate icon to stop the program. 44

45 Summary: In this lab, you learned the mechanism of Sametime Telephony Presence adaptor and leaned how to publish the telephony status on Sametime community and show on Sametime Connect client. The current release of IBM Sametime Unified Telephony Lite Client is not equipped the integrated telephony presence feature yet, and it was planed to be implemented in future release of the product. But using the Sametime Telephony Presence capability which will leverage the Sametime Community Server Toolkit technology, you can add telephony presence in the current product. The telephony presence adaptor is not the feature available only on SUT or SUT Lite products, you can add telephony presence on Sametime Standard environment too. 45

46 IBM Corporation All Rights Reserved. The information contained in this publication is provided for informational purposes only. While efforts were made to verify the completeness and accuracy of the information contained in this publication, it is provided AS IS without warranty of any kind, express or implied. In addition, this information is based on IBM s current product plans and strategy, which are subject to change by IBM without notice. IBM shall not be responsible for any damages arising out of the use of, or otherwise related to, this publication or any other materials. Nothing contained in this publication is intended to, nor shall have the effect of, creating any warranties or representations from IBM or its suppliers or licensors, or altering the terms and conditions of the applicable license agreement governing the use of IBM software. References in this presentation to IBM products, programs, or services do not imply that they will be available in all countries in which IBM operates. Product release dates and/or capabilities referenced in this presentation may change at any time at IBM s sole discretion based on market opportunities or other factors, and are not intended to be a commitment to future product or feature availability in any way. Nothing contained in these materials is intended to, nor shall have the effect of, stating or implying that any activities undertaken by you will result in any specific sales, revenue growth or other results. IBM, the IBM logo, Lotus, Lotus Notes, Notes, Domino, WebSphere and Lotusphere are trademarks of International Business Machines Corporation in the United States, other countries, or both. 46

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