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Configuring CUCM for integration with Pexip Infinity

This section lists the tasks required to configure CUCM so that it can be integrated with one Pexip Infinity location. Each task is described in full in the sections that follow:

This guide is based on CUCM 9.x. If you are using other versions of CUCM, you will need to perform the same set of tasks, but the CUCM menus and options may differ slightly from those described here.

Uploading TLS certificates (TLS only)

If you are using TLS, you need to ensure that CUCM trusts the certificates installed on the Pexip Infinity Conferencing Nodes.

To configure CUCM to trust the Conferencing Node's certificates:

  1. From the CUCM web interface, from the top right Navigation box select Cisco Unified OS Administration and log in to the OS Administration interface.
  2. Go to Security > Certificate Management. and select Upload Certificate/Certificate chain.
  3. In the Certificate Name field, select CallManager-trust.
  4. Upload the certificate of the CA that signed the Conferencing Node's TLS certificate.

Repeat this process, if necessary, to ensure that every Conferencing Node certificate is trusted.

Creating a SIP trunk security profile (TLS only)

If you are using TLS, you must set up a SIP Trunk Security Profile. This is used to configure the TLS parameters for the SIP trunk to Pexip Infinity. For TCP-only (i.e. non-secure) trunks you can use the existing Non-Secure SIP Trunk Profile.

To set up a SIP Trunk Security Profile:

  1. From the CUCM web interface, from the top right Navigation box select Cisco Unified CM Administration and log in.
  2. Go to System > Security > SIP Trunk Security Profile and select Add New.
  3. Complete the following fields:
    NameEnter a name for the profile. This example uses Pexip Infinity Encrypted TLS SIP trunk.
    Device Security ModeSelect Encrypted.
    Incoming Transport TypeSelect TLS.
    Outgoing Transport TypeSelect TLS.
    X.509 Subject NameEnter the X509 subject name of the certificate installed on the Pexip Infinity Conferencing Node (usually the FQDN of the Conferencing Node).
    Incoming PortEnter 5061.
  4. Select Save.

Creating a SIP profile for Pexip Infinity (optional)

CUCM version 9.x includes a preconfigured SIP profile — Standard SIP Profile For Cisco VCS — which you can use for Pexip Infinity integration.

If you want a distinct profile for Pexip Infinity you can copy the Standard SIP Profile For Cisco VCS and set the name, for example, to SIP profile for Pexip Infinity.

Creating a SIP trunk

This configures the connection towards the Pexip Infinity Conferencing Nodes to allow CUCM to make outgoing calls towards Pexip services.

  1. From the CUCM web interface, from the top right Navigation box select Cisco Unified CM Administration and log in.
  2. Go to Device > Trunk and select Add New.
  3. Complete the following fields:

    Trunk Type Select SIP Trunk.
    Device Protocol Select SIP.
    Trunk Service Type Select None (Default).
  4. Select Next.
  5. Complete the following fields (leave all other settings as default):

    Device information
    Device Name Enter a name for the SIP Trunk. This example uses Pexip_Infinity (no spaces are allowed).
    Device Pool Select the pool you want the Conferencing Node to belong to (in CUCM, these are configured in System > Device Pool).
    Location Select the location you want the Conferencing Node to belong to (in CUCM these are configured in System > Location).
    SRTP Allowed Select this option if you are using TLS.
    Call Routing Information > Outbound Calls
    Calling Party Transformation CSS Select <None>.
    SIP Information
    Destination Address is an SRV Select this option if the address you will enter in the field below is an SRV.
    Destination Address

    Enter the address of the Conferencing Node. This can be an IPv4 address, a hostname, or (if the field above is selected) an SRV e.g. example.com.

    You can specify multiple destination addresses if you have more than one Conferencing Node. You do not have to route every node — we recommend just 2 or 3 nodes for redundancy and load sharing purposes.

    Destination Port Enter 5060 for TCP or 5061 for TLS.
    SIP Trunk Security Profile

    If you are using TCP, select the existing Non-Secure SIP Trunk Profile.

    If you are using TLS, select the name of the security profile you have just created. This example uses Pexip Infinity Encrypted TLS SIP trunk.

    SIP Profile Select either Standard SIP Profile For Cisco VCS, or if you have created a new profile — for example SIP profile for Pexip Infinity — select that one instead.
    Normalization Script Select vcs-interop, if available. (This is only necessary if SRTP will be used.)
  6. Select Save

Configuring the dial plan for calls from CUCM to Pexip Infinity

CUCM can be configured to route calls to a SIP trunk based on a specific prefix.

In our example deployment, all our Virtual Meeting Rooms have an alias with a prefix of 555 followed by 3 digits. To ensure that all calls from CUCM-registered devices to Pexip Infinity Virtual Meeting Rooms are routed correctly we must configure CUCM to route calls starting with 555 to Pexip Infinity without changing the number that was dialed.

To route calls based on a specific prefix:

  1. From the CUCM web interface, from the top right Navigation box select Cisco Unified CM Administration and log in.
  2. Go to Call Routing > Route/Hunt > Route Pattern and select Add New.
  3. Complete the following fields:

    Pattern Definition
    Route Pattern Enter 555XXX
    Description Enter a description of the pattern. This example uses Route 555XXX to Pexip Infinity SIP trunk.
    Gateway/Route List Select the SIP trunk you created earlier. This example uses Pexip_Infinity.
    Call Classification Select OnNet.
    Provide Outside Dial Tone Do not select this option.
  4. Select Save.

Adding a SIP route pattern

SIP route patterns are used to route calls based on a SIP destination FQDN rather than a numeric route pattern.

  1. From the CUCM web interface, from the top right Navigation box select Cisco Unified CM Administration and log in.
  2. Go to Call Routing > SIP Route Pattern and select Add New.
  3. Complete the following fields (leave all other settings as default):

    Pattern Usage Select Domain Routing.
    IPv4 Pattern Enter the domain to route. This example uses example.com.
    SIP Trunk Select the SIP trunk you created earlier. This example uses Pexip_Infinity.
  4. Select Save.