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Direct SIP: Cisco Unified Communications Manager 6.1

Communications Server 2007 R2

If you’re looking for an introduction about how to configure Cisco Unified Communications Manager (CUCM) version 6.1.2 with Office Communications Server 2007 R2, this article will help you. The proper way to connect and interoperate the two products is to set up a Direct SIP connection in Office Communications Server. On the Cisco side, this is referred to as a SIP trunk. This article covers how to configure a SIP trunk in CUCM.

Authors: Jerome Berniere, Rui Maximo

Publication date: December 2009

Product version: Office Communications Server 2007 R2

If you’re looking for an introduction about how to configure Cisco Unified Communications Manager (CUCM) version 6.1.2 with Office Communications Server 2007 R2, this article will help you. The proper way to connect and interoperate the two products is to set up a Direct SIP connection in Office Communications Server’s parlance. In Cisco’s parlance, this is referred to as a SIP trunk. So, when discussing the configuration of CUCM, we’ll refer to this interop connectivity as SIP trunk. When explaining the configuration of the Office Communications Server, we’ll refer to this interop connectivity as a Direct SIP.

If you’re trying to operate CUCM with Office Communications Server 2007 R2, you probably already have CUCM deployed in your environment for all your IP phones and your CUCM configured to route all external calls to the PSTN through a PSTN trunk. In this case, you’ll likely want to configure Office Communications Server 2007 R2 Enterprise Voice to route external calls to the PSTN through the CUCM PSTN trunk. Calls between Communicator users and Cisco IP phone users should be possible using each user’s unique extension number, and users are accessible via externally routable direct inward dialing (DID). This is illustrated in Figure 1.

Figure 1. Routing of calls between Communicator users and Cisco IP phone users

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There are two parts to interoperating the two products: configuring CUCM, and configuring Office Communications Server. The high level steps for configuring CUCM are as follows:

  1. Create partition

  2. Create calling search space

  3. Define translation patterns

  4. Provision SIP trunk

  5. Set up route pattern

The following sections describe the CUCM configuration steps in more detail. For details about how to configure Direct SIP on Office Communications Server 2007 R2 Mediation Server, see Direct SIP: Configuring Mediation Server.

Let’s dive into the configuration steps for CUCM. The connectivity between CUCM and the Mediation Server is referred to as SIP trunk to conform to Cisco’s terminology.

A CUCM partition contains a list of route patterns. Partitions facilitate call routing by dividing the route plan into logical subsets that are based on organization, location, and call type. For details about partitions, see "Partitions and Calling Search Spaces" in the CUCM System Guide. For a demonstration of how to create a partition in CUCM, see Desmond Lee’s video.

This partition is specific to the SIP trunk that connects Office Communications Server to CUCM. It will store route pattern and translation pattern rules that are specific to handling incoming traffic to CUCM from the Mediation Server. For our example, this partition is named, "OCSIncoming".

To create a CUCM partition
  1. From your CUCM Administration console, click Call Routing, click Class of Control, and then click Partition (Figure 2).

    Figure 2. Opening the partitions screen

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  2. Click Add New to create a new partition as showing in Figure 3.

    Figure 3. Adding a new partition

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  3. In the Name field, specify the name of the partition and a description as shown in Figure 4. We’ve specified "OCSIncoming, OCSIncoming".

    Figure 4. Name and describe the partition

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  4. Click Save, and then click Find to verify that the partition was created. The new partition should appear in the result set as shown in Figure 5.

    Figure 5. New partition successfully created

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  5. To view the configuration, click the new partition name (Figure 6).

    Figure 6. Partition information

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    noteNote:
    Whether a partition is required or not in a particular environment may depend on the rules required. Please verify with your CUCM administrator.

A CUCM calling search space is an ordered list of route partitions. Calling search spaces determine which partitions and the order in which they are searched when CUCM routes a call. In our example, this calling search space is named "OCSIncoming" and is assigned to the OCSIncoming partition created earlier. For a demonstration of how to create a calling search space in CUCM, see Desmond Lee’s video.

To create a calling search space
  1. From the CUCM Administration console, click Call Routing, click Class of Control, and then click Calling Search Space (Figure 7).

    Figure 7. Opening the Calling Search Space Configuration screen

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  2. Click Add New to create a new calling search space. Specify a name for the calling search space. In our example, the new calling search space is "OCSIncoming".

  3. Select the OCSIncoming partition, and then click the down arrow to move it to the Selected Partitions area to assign this calling search space to the OCSIncoming partition (Figure 8).

    Figure 8. Assigning the OCSIncoming calling search space to the OCSIncoming partition

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  4. Click Save to insert this calling search space in the CUCM configuration.

Next, translation patterns must be defined and assigned to the newly created partition. Translation patterns manipulate dial strings before a call is routed.

The translation patterns we’ll define are specific for handling inbound calls to the CUCM that come from Office Communications Server. A translation pattern converts dial strings for calls sent by the Mediation Server where the TO field matches the pattern. CUCM then uses the translated dial string to determine how to route the call. For a demonstration of how to define translation patterns in CUCM, see Desmond Lee’s video.

One or more translation patterns must be defined depending on your needs. At a minimum, you need to define at least two translation patterns:

  • A translation pattern for incoming calls from Office Communications Server that are destined for the PSTN. This will allow Office Communicator users who are enabled for Enterprise Voice to dial out externally.

  • A translation pattern for incoming calls from Office Communications Server that are destined for CUCM assigned phone numbers. This will allow Office Communicator users to dial Cisco phones within the organization.

If users are allowed to dial international numbers, you will have to define a third translation pattern. Consider creating translation patterns for emergency numbers (for example, 911).

Office Communications Server to PSTN Translation Pattern

The first translation pattern (called 33.xxxxxxxxx, which means the TO string starting with 33 followed by 9 digits) transforms all calls from Office Communications Server that are destined to a domestic PSTN number.

On your CUCM Administration console, click Call Routing, click Translation Pattern (Figure 9), and then click Add New to create a new translation pattern configuration.

Figure 9. Opening the Translation Pattern Configuration screen

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The translation pattern is assigned to the OCSIncoming partition that you created.

The period (.) is a separator that indicates that all digits before the period are to be discarded. This separator is used in the Called Party Transformations section (see Figure 10) when the Discard Digits setting is set to PreDot. In our example, 33 will be removed from the dial string.

For the FROM field, the last 9 digits are retained. The country prefix (which is 33 for France) is stripped from the E.164 calling party dial string and is presented to the PSTN in the required format, 16986xxxx. In this case, the translation pattern transforms the dial string from 3316986xxxx to 16986xxxx.

For the TO field, the pattern strips the leading 33 and adds 00 as a prefix to the remaining string. The translation pattern transforms the dial string from 33xxxxxxxxx to 00xxxxxxxxx, where the first 0 is the outside line prefix that is required to obtain an outside line on CUCM.

Figure 10. Configuring a new translation pattern

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Office Communications Server to CUCM Translation Pattern

The second translation pattern (in this example, 3316986xxxx) transforms all calls from Office Communications Server that are destined for a CUCM assigned number. The translation pattern is assigned to the OCSIncoming partition that you created earlier. Because all calls destined from Office Communications Server will be in RFC 3966 format, calls in this example are prefixed by the country code, 33 for France, and then followed by the prefix for all numbers that are assigned to Cisco IP phones, 16986, in the enterprise. This pattern is applied to calls where the TO field matches the pattern, 3316986xxxx. Because CUCM selects translation patterns on the basis of the best match, this translation pattern will be selected instead of the first translation pattern.

In our example, this translation pattern translates dial strings for calls sent by the Mediation Server where the TO field matches the pattern, 3316986xxxx. It strips all leading digits from TO and FROM fields and retains only the last 4 digits, xxxx. Dial strings with the pattern 3316986xxxx are translated to dial strings of the form xxxx. As shown in Figure 11, this translation is performed on the callee’s (TO field) number and the caller’s (FROM field) number.

Figure 11. Office Communications Server to CUCM translation pattern

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A SIP trunk to the Mediation Server is created on the CUCM, and is assigned the OCSIncoming calling search space that was created earlier. The SIP trunk uses SIP over TCP, which requires port 5060.

The CUCM SIP trunk is established to the IP address of the Mediation Server that represents the gateway listening IP address. The Mediation Server must be configured with two IP addresses. One IP address connects to the Office Communications Server infrastructure; the other IP address connects to the CUCM, which represents the gateway.

If you’re not sure which IP address is the gateway listening IP address for your Mediation Server, see Figure 4 in the article Direct SIP: Configuring Mediation Server.

In our example, the trunk name is Trunk_to_OCS, and the Mediation Server’s gateway listening IP address is 192.168.0.105. For a demonstration of how to provision a SIP trunk in CUCM, see Desmond Lee’s video.

To create a new trunk
  1. On the CUCM Administration console, click Device, and then click Trunk (Figure 12).

    Figure 12. Adding a new trunk

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  2. Click Add New, and then on the Trunk Type list, select SIP Trunk. Select SIP as the Device Protocol, and then click Next (Figure 13).

    Figure 13. Configuring the SIP trunk

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  3. Specify the Device Name (in our example, Trunk_to_OCS), the Calling Search Space (in our example, OCSIncoming), the Rerouting Calling Search Space (in our example, OCSIncoming), the Destination Address to match the Mediation Server gateway listening IP address (in our example, 192.168.0.105), and Destination Port (in our example, 5060).

    Table 1 lists the required Trunk Configuration settings and their values (Figure 14).

    Table 1. Required Trunk Configuration settings

    Setting

    Value

    Device Pool

    Default

    Call Classification

    Use system default

    Media Resource Group List

    MRGL1

    Location

    Hub_None

    Packet Capture Mode

    None

    Media Termination Point Required

    Select check box

    Retry Video Call as Audio

    Select check box

    Significant Digits

    All

    Connected Line ID Presentation

    Default

    Connected Name Presentation

    Default

    Calling Party Selection

    Originator

    Calling Line ID Presentation

    Default

    Calling Name Presentation

    Default

    MTP Preferred Originating Codec

    711ulaw

    Presence Group

    Standard Presence group

    SIP Trunk Security Profile

    Non Secure SIP Trunk Profile

    SIP Profile

    Standard SIP Profile

    DTMF Signaling Method

    No Preference

    Figure 14. Trunk Configuration settings

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  4. Click Save, and then reset the trunk by clicking the Reset icon at the top of the Trunk Configuration page to activate the SIP trunk changes.

Route patterns are designed for routing calls from CUCM to the Mediation Server over the SIP trunk so that users who are using a Cisco IP phone can call Office Communicator users. The route patterns determine which calls are sent to the SIP trunk based on a pattern match of the phone number in the TO field. Route patterns can also perform transformations of the dial strings for the TO and the FROM fields. For a demonstration of how to setup a route pattern in CUCM, see Desmond Lee’s video.

From the CUCM Administration console, click Call Routing, click Route/Hunt, and then click Route Pattern (Figure 15). Click Add New.

Figure 15. Opening the Route Pattern screen

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For example, if you want users who are using a Cisco IP phone to be able to dial an Office Communicator user using a 4-digit extension, create a route pattern that is associated to the Trunk_to_OCS SIP trunk that you created earlier that instructs CUCM to route to the Mediation Server all calls that match the TO dial string with the pattern xxxx. In this case, no transformation of the dial strings in the TO and the FROM fields is necessary. This route pattern configuration is illustrated in Figure 16.

Figure 16. Route pattern

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And that completes your introduction to configuring a SIP trunk on Cisco’s Unified Communications Manager (also known as CallManager).

To learn more, check out the following articles:

 
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