Enabling moderator dial out and joining with device

You can enable a meeting moderator to dial out to an external user and enable an IBM® Sametime® user to join the same audio video conference with their external device which can be a SIP device or a mobile phone. Both features are enabled when you use this procedure to enable telephony on the Sametime Conference Manager.

Procedure

  1. Browse to the WebSphere® Integrated Solutions Console on the server hosting the Sametime System Console, http://ssc_host:8700/ibm/console.
  2. Log in to the Integrated Solutions Console as the WebSphere administrator.
  3. Click Sametime System Console > Sametime Servers > Sametime Video Manager Server.
  4. Select the Sametime Video Manager Server you want to work with.
  5. Click Configure the Sametime SIP peer.
  6. Enter the Sametime SIP Proxy and Registrar host name and description in the Name and Description fields.
  7. Enter the TLS Port number 5081.
  8. Select the Use route header option, and in the Transport type field, select TLS.
    Note: Ensure that the Transport Protocol setting in the SIP Proxy Registrar section of the Sametime Media Manager settings is the same as the Transport type you specified in this step.
  9. Click OK.
  10. Use this format for the recommended SIP URI for dialing out:

    sip_schema:userid@ipddr/fqdn:sip_port;transport=transport_protocol

    For example:

    sip:alice@sipproxyhost.xyz.com:5080;transport=tcp

    sips:alice@sipproxyhost.xyz.com:5081

    Note: To simplify the SIP URI for dialing out, use routing rules which accept the request URI, for example, sip:2343434343@abc.xyz.com and convert it to sip:2343434343@abc.xyz.com:5080;transport=tcp.
  11. To create the routing rule, click Sametime System Console > Sametime Servers > SIP Proxies and Registrars.
  12. Select the server you want to work with, and then click Proxy Administration.
  13. Click New. Use these parameters to complete the fields:
    Note: In the parameters for the new rule, replace abc.xyz.com with your own external SIP gateway in each instance.
     <routeRules>
 <rule priority="1" description="" name="DialOut">
 <condition type="method"><![CDATA[INVITE]]></condition>
 <condition type="requestURI"><![CDATA[sip:.*@abc.xyz.com]]></condition>
	 <destination>
	 <output>
	 <inputPattern type="requestURI" value="sip:(.*)@.*"/>
	 <outputPattern type="requestURI" value="sip:$1@abc.xyz.com:5080;transport=tcp"/>
	 </output>
	 </destination>
	 </rule>
	 </routeRules>
  14. Click OK.
  15. Resynchronize the nodes by completing these steps:
    1. In the Deployment Manager's Integrated Solutions Console, click System Administration > Nodes.
    2. In the Nodes table, select all nodes in the cluster.
    3. Click Full Resynchronize.
  16. Restart the SIP Proxy and Registrar.
  17. On the Sametime Conference Manager, navigate to

    {WAS_INSTALL_ROOT}/profiles<cf_profile>/installedApps/cellName/ConferenceFocus.ear/ConferenceFocus.war

    Edit the ConferenceManager.properties file and set TelephoneConferenceEnabled=true.

  18. Save and close the ConferenceManager.properties file.
  19. If there is one Sametime Conference Manager, restart it now. If you deployed a cluster of Sametime Conference Managers, synchronize all nodes in the cluster as follows.
    1. In the Deployment Manager's Integrated Solutions Console, click System Administration > Nodes.
    2. In the Nodes table, select all nodes in the cluster.
    3. Click Full Resynchronize.