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
- Browse to the WebSphere® Integrated
Solutions Console on the server hosting the Sametime System Console,
http://ssc_host:8700/ibm/console
. - Log in to the Integrated Solutions Console as the WebSphere administrator.
- Click Sametime System Console > Sametime Servers > Sametime Video Manager Server.
- Select the Sametime Video Manager Server you want to work with.
- Click Configure the Sametime SIP peer.
- Enter the Sametime SIP Proxy and Registrar host name and description in the Name and Description fields.
- Enter the TLS Port number 5081.
- 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.
- Click OK.
- 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 tosip:2343434343@abc.xyz.com:5080;transport=tcp
. - To create the routing rule, click Sametime System Console > Sametime Servers > SIP Proxies and Registrars.
- Select the server you want to work with, and then click Proxy Administration.
- 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>
- Click OK.
- Resynchronize the nodes by completing these steps:
- In the Deployment Manager's Integrated Solutions Console, click System Administration > Nodes.
- In the Nodes table, select all nodes in the cluster.
- Click Full Resynchronize.
- Restart the SIP Proxy and Registrar.
- 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 setTelephoneConferenceEnabled=true
. - Save and close the
ConferenceManager.properties
file. - 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.
- In the Deployment Manager's Integrated Solutions Console, click System Administration > Nodes.
- In the Nodes table, select all nodes in the cluster.
- Click Full Resynchronize.