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The integration between PrivateServer Appliance and the Cisco Unified Call Manager Business Edition version 8.6.2 (from now on shortly CUCMBE) is made possible by setting up a SIP Trunk. Both the appliances can speak SIP, both work out the UDP transport protocols and the G711/ulaw codec and that's mostly all we need to start.

1. Configuring the CUCMBE

As the CUCMBE is prone to a wide range of configurations and network VoIP setup, we will explain to you the basic configuration and let the CUCMBE managers work out how to integrate it in their own environment. It's out of the scope of the present documentation to teach you how to manage a CUCMBE.

1.1 Prerequisite

Of course you need the CUCMBE appliance, which may come both in VMWare virtualization fashion or in real steel installed one. In either cases we assume that:

  • basic services are activated (see the fig. X). Beware: the Cisco IP Voice Media Streaming App 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

1.2 Basic CUCMBE configuration

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

 

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