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basic services are activated (see the fig. X). Beware: the "Cisco IP Voice Media Streaming App" which service must be activated in order to create an MTP (Media Termination Point)
- The phones (both SCCP or SIP ones) are correctly configured and functioning
- Your network design let the UDP packets to stream from the PrivateServer to the CUCMBE and vice versa
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If the prerequisites are matched then we first need to configure a new SIP Trunk. Go to the Cisco Unified CM Administration and from its GUI use the Device->Trunk menu entry to create a new one.
First select The Trunk Type to SIP Trunk and the Device Protocol to SIP, as in fig. X. The Trunk Service Type can be None.
After you press the Next button, you got a long page of configuration to deal with. We are going to give you the essential configuration parameter to make the trunk work with PrivateServer, so you can keep the default options configuration if not else stated.
You can choose your own Device Name and Description. Please set the Device Pool to the one containing the phones you actually want to put in contact with PrivateServer. Also check the Media Termination Point Required option. We also check the Retry Video Call as Audio for the possible Video phone (PrivateServer doesn't support video calls). Check the PSTN Access and the Run On All Active Unified CM Nodes. The rest of the options can stay in default values, till the SIP Information form (see fig. X), just pay attention to configure the Calling Search Space accordingly with your design.
In the SIP Information form we