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QUANTiX QFlex Session Border Controller

CUCM 10.0 - XO SIP Trunk Configuration Guide

Release 5.6.2-9

Document revision: 01.01

www.genband.com

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5/23/2014 CUCM 10.0 - XO SIP Trunk Configuration Guide 630-02102-01 01.01

Copyright © 2014 GENBAND. All Rights Reserved.

630-02102-01

QUANTiX QFlex Session Border Controller Publication: 630-02102-01 Document status: Standard Document release date: 5/23/2014 Copyright © 2014 GENBAND. All rights reserved. Use of this documentation and its contents is subject to the terms and conditions of the applicable end user or software license agreement, right to use notice, and all relevant copyright protections. GENBAND, the GENBAND corporate logo and tagline, and certain of GENBAND's product and solution names are registered trademarks of GENBAND and its affiliates. While the information in this document is believed to be accurate and reliable, except as otherwise expressly agreed to in writing, GENBAND AND/OR ITS LICENSORS PROVIDES THIS DOCUMENT "AS IS" WITHOUT WARRANTY OR CONDITION OF ANY KIND, EITHER EXPRESS OR IMPLIED. The information and/or products described in this document are subject to change without notice. THE INFORMATION CONTAINED HEREIN IS THE PROPERTY OF GENBAND OR ITS LICENSORS AND MUST NOT BE DISCLOSED, OTHER THAN TO EMPLOYEES OF GENBAND OR CUSTOMERS WITH NEED-TO-KNOW, WITHOUT THE PRIOR WRITTEN CONSENT OF GENBAND OR ITS LICENSORS. THE INFORMATION MUST NOT BE DISCLOSED TO SUBCONTRACTORS OR REGULATORY AUTHORITIES. GENBAND MUST BE NOTIFIED OF ANY REQUEST OR ORDER FOR DISCLOSURE PRIOR TO SUCH DISCLOSURE.

Document Revision History

Document Revision 01.01 5/23/2014

This is the first release of the document.

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Table of Contents

1 Preface 6

1.1 Audience 6

1.2 Introduction 6

1.3 Configuration Data 6

1.4 System Integration Caveats and Notes 6

Caveats 6

Configuration Notes 6

1.5 System Integration and Load Lineup 7

2 System Overview 8

2.1 Network Diagram 8

2.2 Load Lineup 8

3 Cisco Unified Communications Manager (CUCM) 9

3.1 SIP Trunk Security Profile 9

3.2 SIP Profile 11

3.3 Trunk 14

PSTN Trunk 14

Unity Connection Trunk 21

3.4 Voice Mail 26

Voice Mail Pilot 26

Voice Mail Profile 27

3.5 Phones 29

SIP Phone Security Profile 29

Add SIP Phone 30

SCCP Phone Security Profile 37

Add SCCP Phone 38

3.6 FAX 49

3.7 Route Patterns 55

Route Pattern to PSTN 55

Route Pattern to Cisco Unity Connection 58

4 QFlex 60

4.1 Create New Service Configuration 61

Trunk Provisioning 63

SIP Management 82

Platform 86

4.2 RTP Relay Configuration 87

Interface Pairs 88

Control Access 89

Parameters 90

4.3 Server Configuration 92

Add LAN-side Static Route 92

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IP Address White List 93

5 Cisco Unity Connection 95

5.1 Create CUCM Integration 95

5.2 Create a User 96

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1 Preface

1.1 Audience This document is intended for GENBAND’s QFlex technical staff and any individual tasked with the integration of GENBAND’s QFlex enterprise SBC with a Cisco Unified Communications Manager (CUCM) and XO Communication’s SIP Trunking service.

1.2 Introduction Configuration Guides are written to provide a system integrator the necessary information to construct an integration similar to what has been built for certification or IOT purposes.

This document’s focus is on the certification of the GENBAND QFlex SBC along with Cisco Unified Communications Manager (CUCM) version 10 with XO Communication’s SIP Trunking service. The reference architecture is shown in Figure 1.

1.3 Configuration Data The configuration data provided within is shown in graphic user interface (GUI) format where possible. The appropriate data values are shown graphically, largely in their default values. Comments will be provided alongside many GUIs to enforce/explain various parameter values to be provisioned.

Some configuration will be provided in command line interface where applicable.

1.4 System Integration Caveats and Notes

Caveats

• Caller ID on a call transfer target device in the PSTN shows the transferor ID, not the transferee. Observed in both ringing and consultative call transfers.

Configuration Notes

• For full compliance of T.38 test cases, XO’s network was configured to force fax calls to T.38. Note that G.711 fax works in this test environment when XO’s network is configured for G.711 Fax / T.38 Fax.

• REFER-based call transfer requests from CUCM are not supported by QFlex. All Call Transfer functionality passed using INVITE-based call transfer method.

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1.5 System Integration and Load Lineup This testing effort measures the functionality of a SIP Trunking integration between XO Communications and a Cisco UCM with GENBAND QFlex eSBC.

System component software/firmware versions are captured below.

Component Vendor Model Version

IP-PBX Cisco CUCM 10.0.1.11900-2

SBC GENBAND QFlex 5.6.2-9

Phones Cisco 7960 P0030801SR02

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2 System Overview

2.1 Network Diagram

Figure 1 - System Diagram

2.2 Load Lineup

Component Vendor Model Version

IP-PBX Cisco CUCM 10.0.1.11900-2

SBC GENBAND QFlex 5.6.2-9

Phones Cisco 7960 P0030801SR02

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3 Cisco Unified Communications Manager (CUCM)

This section presents the CUCM configuration items that were provisioned in the integration with QFlex for SIP Trunking service with XO Communications.

3.1 SIP Trunk Security Profile This profile will be referenced when creating the actual SIP Trunk that connects CUCM to QFlex. Browse to “System > Security > SIP Trunk Security Profile” and create a profile as seen in the graphic below.

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• “Device Security Mode set to Non Secure

• “Incoming Transport Type” set to TCP+UDP

• “Outgoing Transport Type” set to UDP

• “Incoming Port” set to 5060

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3.2 SIP Profile This profile will be referenced when creating the actual SIP Trunk that connects CUCM to QFlex. Browse to “Device > Device Settings > SIP Profile and complete the profile configuration as seen below.

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3.3 Trunk A SIP trunk connects CUCM to an external SIP server. Access the CUCM Trunk Configuration menu from “Device > Trunk” menu.

PSTN Trunk

This trunk carries CUCM client phone calls to/from the PSTN via the QFlex SBC.

Create the QFlex trunk as seen in Figure 2.

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Figure 2 – Add New Trunk

Click the “Next” button and complete the configuration details as seen in the graphics below.

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• Provide a descriptive name to reference the SIP Trunk in the “Device Name” field.

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• Enable “Asserted-Identity” and set “Asserted-Type” to PAI.

• “SIP Privacy” set to Default. Set this parameter to “ID” when CUCM is to provide full Privacy signaling in outbound calls.

• “Inbound Calls > Significant Digits” set to 4 as this is the CUCM subscribers extension digit length.

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• Under “SIP Information” provide the QFlex LAN-side IP Address and its SIP Listen Port in the proper Destination fields.

• Reference the “SIP Trunk Security Profile” which represents the transport method employed by the SIP Trunk as seen in section 3.1.

• Reference the “SIP Profile” as configured in section 0.

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Unity Connection Trunk

This SIP trunk connects CUCM to the Unity Connection Voice Mail Server.

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• Provide a descriptive “Device Name” since this trunk will be referenced later.

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• Under “Destination”, enter the Cisco Unity Connection IP Address and Port.

• Reference the non Secure “SIP Trunk Security Profile”.

• Reference the standard “SIP Profile”.

3.4 Voice Mail Configure the following in CUCM to setup access to external Voice Mail with Cisco Unity Connection.

Voice Mail Pilot

Browse to “Advanced Features > Voice Mail > Voice Mail Pilot” and create a pilot number as seen in the graphic below.

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Voice Mail Profile

Browse to “Advanced Features > Voice Mail > Voice Mail Profile”.

Create a Voice Mail Profile as seen in the graphic below. This profile is then referenced in any CUCM User where the voice mail functionality is required.

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• Provide a descriptive “Voice Mail Profile Name” for subsequent use in referencing voice mail functionality.

• Reference the “Voice Mail Pilot” as created above.

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3.5 Phones For this IOT, SCCP phones were used. However, both SCCP and SIP type phone provisioning is discussed in this section. The Cisco 7960 phone supports both SCCP and SIP protocols.

SIP Phone Security Profile

Access “System > Security > Phone Security Profile” and create a UDP-based SIP security profile as seen in the graphic below.

• Set “Transport Type” to UDP.

• Specify port 5060 for “SIP Phone Port”.

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Add SIP Phone

Browse to “Device > Phone” and click the “Add New” button at bottom.

• From the pull-down window, specify the “Phone Type” corresponding to your hardware and hit the “Next” button.

• From the pull-down window, specify the protocol to be used between the phone and CUCM. In this case, it’s SIP.

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• In the “MAC Address” field, provide the MAC address that is printed on the bottom/back of the phone.

• “Device Pool” set to Default.

• “Phone Button Template” set to reference that of physical phone used here.

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• Provide the appropriate “Device Security Profile” that references the profile as discussed under “Phone Security Profile” above.

• Reference the “SIP Profile” as discussed in section 0.

Click the Save button to save your Phone entry.

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Line Information

• Click the “Line [1] – Add a new DN” link.

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• Provide the desired extension value in the “Directory Number” field.

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SCCP Phone Security Profile

Access “System > Security > Phone Security Profile” and create a SCCP security profile as seen in the graphic below.

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Add SCCP Phone

Browse to “Device > Phone” and click the “Add New” button at bottom.

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Device Information

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• Reference the SCCP Security profile created earlier in the “Device Security Profile” field.

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Line Information

• Provide the extension number in the “Directory Number” field.

• Reference the “Voice Mail Profile” created in section 3.4.

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• Configure Call Forwarding targets in “Destination” field. Options for differentiated CF exists based on caller being internal CUCM extension or external PSTN. If calls are to roll to voice mail, simply check the checkbox under “Voice Mail”.

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• The “Line Text Label” value will be shown on the phone’s display.

• Normalize the external caller ID to 10 digits by provisioning the “External Phone Number Mask” as seen above.

• The Multiple Call / Call Waiting Settings are shown with their default values. Move the “Busy Trigger” to 1 to help with Call Forward Busy testing.

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3.6 FAX A CUCM Device is required to represent the SIP/analog IAD gateway placed between CUCM and the analog FAX machine. Add this device by browsing to “Device > Gateway”. Click the “Add New” button to add the Gateway device.

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• Click the “Save” icon to commit and bring up additional items for consideration as seen below.

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• Click the “Save” icon to commit and bring up additional items for consideration as seen below.

• Click on the analog port icon next to the physical port on the IAD that will be used to connect with the FAX machine. In this example, we use port 0/3/0.

• Select POTS “Port Type” and hit the “Next” button to bring up the provisioning interface as seen below.

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Now add a DN to the gateway configuration to represent the FAX number.

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3.7 Route Patterns A Route Pattern ties a digit pattern dialed by a CUCM client to SIP trunk connecting CUCM to an external service. Specifically, n digits are reserved by the administrator to be dialed in addition to a target phone number on a remote system. The n digits are referred to as the “Route Pattern” in CUCM terminology.

To configure a Route Pattern, browse to “Call Routing > Route/Hunt > Route Pattern” and click the “Find” button to show all existing Routes.

Route Pattern to PSTN

Create a Route Pattern for users to dial as a prepend to the target phone number. This pattern will point to an outbound route for the call to take over a SIP Trunk. In this IOT, the route pattern 8 was used as seen in the Route Pattern below to steer PSTN calls to the SIP trunk connecting the CUCM to the local QFlex and ultimately to the XO SIP Trunking service.

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• In the “Route Pattern” field, the value 8.@ indicates that any phone number dialed with a 8 prefix will be routed out the SIP Trunk referenced in the “Gateway/Route List” field.

• Set “Numbering Plan” to NANP.

• Reference the target SIP Trunk in the “Gateway/Route List” field.

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• The “Calling Party Transform Mask” should be enabled and defined with six digits to complement the four-digit extension range defined for this IOT. This empowers the CUCM to build an outgoing INVITE message with a 10 digit caller ID value.

• In the “Called Party Transformations” window, enter a value of PreDot in the “Discard Digits” field to trigger CUCM to discard the Route Pattern identifier digit dialed by users making PSTN calls.

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Route Pattern to Cisco Unity Connection

• Provide a voice mail pilot number in the “Route Pattern” field.

• Associate the SIP Trunk which connects CUCM to Cisco Unity Connection in the “Gateway/Route List” field.

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4 QFlex

Access the QFlex Server via the QFlex Element Management System (EMS) web-based provisioning management interface at http://<EMS IP Address>.

QFlex EMS

Login with an account that has administrator privileges.

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4.1 Create New Service Configuration Create a new service configuration by selecting “Configuration > Service Configuration” from the QFlex menu and then clicking on the “Create Service Configuration” button as seen below.

Give the new Service Configuration a descriptive name and click the “Create” button.

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The default service configuration (previously configured) will appear as seen in the graphic below. Edit this configuration to fit the IOT environment as discussed below.

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Trunk Provisioning

Edit LAN Logical Interface

Provide the QFlex LAN-side (i.e., PBX side) interface attributes as seen in Figure 3.

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Figure 3 – LAN-side Interface Configuration

• Provide the LAN-side QFlex IP Address in the “Address” field.

• Select the physical QFlex “Interface” which supports the LAN-side connection.

• Provide the “Netmask” associated with the assigned IP Address.

Click the “Update” button to save the configuration.

Edit WAN Logical Interface

Repeat the above procedure for the QFlex WAN-side Logical Interface as seen in Figure 4.

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Figure 4 – WAN-side Interface Configuration

Now define the Application Interfaces. This is where QFlex is configured with the adjacent nodes’ IP interfaces information and the mechanisms which govern how the communication is to occur. To edit an application interface, click on the ‘Edit’ hyperlink as seen in the graphic below.

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Edit LAN Application Interface

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• Reference the corresponding “Logical Interface” as defined earlier for the LAN side.

• The “Mate Application Interface” represents the LAN side’s paired interface, i.e., the WAN side interface.

• Select the “PBX/Gateway Profile” which mostly closely identifies the PBX.

• Indicate the transport protocol to be used between QFlex and the PBX by specifying the QFlex listen port in the proper protocol field. Values of 0 mean that transport is disabled.

• In the “Network” field, select Local Area Network.

Add a “Next Hops” entry for each CUCM node in the CUCM cluster that will be interacting with QFlex. Be sure to check your CUCM cluster setup to understand how many nodes are deployed.

Specify the IP Address, Transport Type and Listen port for each PBX node. Note that a value of “0” effectively deactivates a transport type. When finished, click the “Update” button to save the provisioned data.

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Next, the following “Advanced Parameters” will be shown for any necessary modification.

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Edit WAN Application Interface

• Reference the corresponding “Logical Interface” as defined earlier for the WAN side.

• The “Mate Application Interface” represents the paired interface, i.e., the LAN side interface.

• Select the “PBX/Gateway Profile” SIP BroadsoftBW 1 as it best represents the SIP entity within XO’s SIP Trunking service.

• Indicate the transport protocol to be used between QFlex and XO by specifying the QFlex listen port in the proper protocol field. Values of 0 mean that transport is disabled.

• In the “Network” field, select Wide Area Network.

Add the “Next Hop” Network info representative of the XO Service as seen below.

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• Provide the IP address of the XO Signaling Server in the “Address” field.

• Specify the associated XO listen port in the field which represents the transport protocol to be used between QFlex and XO. Note that a value of 0 effectively deactivates a transport type.

Click the “Add” button to add the Next Hop data.

Click the “Update” button (at the bottom of the page) to save the WAN Application Interface data.

Next, the following “Advanced Parameters” will be shown for any necessary modification.

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• Set “DTMF Type” to 2833 Event Packets for RFC 2833 DTMF support across the WAN link. Set this parameter to Audio RTP Packets to force QFlex to use in-band DTMF.

• Use the “Prepend Digits on Diversion Header” field to control if and how QFlex prepends digits to Diversion headers within SIP requests received from the LAN side and transmitted to the WAN side. In the CUCM integration with XO, CUCM populates the Diversion header (in use cases such as call forwards to the PSTN) with the CUCM extension value – in this case a 4-digit extension. XO requires a 10 digit Diversion header representative of one of its DNs. Hence, this QFlex field is used to augment the received Diversion header to its full 10-digit DN value. PLEASE NOTE: This QFlex normalization

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is global in nature, that is, it’s good for one number range. If the enterprise has more than one block of DNs which collectively cannot be represented by a single entry in this QFlex field, then the Diversion normalization must be performed in CUCM using a script which can provide the flexibility to define multiple DN ranges and Diversion header augmentation.

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Trunks

Click the “Add” link next to the Trunks section of the Service Configuration as seen above. This will present the GUI as seen in Figure 5.

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Figure 5 - Trunk Definitions

• Reference both the WAN and LAN application interfaces created earlier.

• Declare a DTMF payload ID of 101.

• Click the “Add” button to confirm.

Now Edit the Trunk as seen in Figure 6. This step allows for further specification of the links to each adjacent network node.

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Figure 6 - Edit Trunk

Under “Audio Codecs”, select the “Add” link.

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Add both the G.711 uLaw and G.711 aLaw codecs and then select the “Update” button to confirm these settings as seen in Figure 7.

Figure 7 - Confirm Codecs

Now associate this entire new configuration with a physical QFlex server. Click the “associate” link as seen in the following graphic.

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Figure 8 - Associate Configuration

Click on the Physical QFlex server name representative of the host QFlex server from the available list as seen Figure 9. If no physical servers are shown, enter “%” in the search field and click the “Find” button.

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Figure 9 - Physical QFlex Servers

Now download the configuration to the selected physical server as seen in Figure 10.

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Figure 10 - Download Configuration

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Figure 11- Download Confirmation

You should see a “QFlex configuration downloaded successfully” message on the resulting GUI presentation.

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SIP Management

Access “Configuration > Service Configuration” and click on “SIP Management” as seen in Figure 12.

Figure 12 - SIP Management

Pilot Number Registration

XO Communications does not support eSBC registrations, so ensure that QFlex has no Pilot Number configured as seen in the graphic below.

The default authorization username, password and refresh rate are auto populated by default but are dormant and can be ignored.

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Session Timers

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QFlex’s support of SIP Session Timers can be managed here per the various test case requirements. In the example below, QFlex is the Session Refresher, sending re-INVITEs at 90 second intervals. This refresh interval is a bit short for production environments, but for testing SIP Session Timer usage, it works quite well. Move this value to 1800 seconds once session refresh test case execution is complete.

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Parameters: SIP Management

Figure 13 - SIP Management

• Enable SIP OPTIONS, if need, by providing a value for the “SIP Heartbeat Interval” field.

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Platform

QFLex’s internal DNS can be used. To enable this DNS, browse to the Service Configuration and click on the “Platform” link as seen in Figure 14.

Figure 14 - Enable Internal DNS

Create a DNS Server with IP address and port as seen below.

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4.2 RTP Relay Configuration Create a new RTP Relay configuration as seen below.

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Interface Pairs

• Provide the QFlex Public and Private side IP addresses as seen above.

• The “Max Tunnels” value must be set to a numeric value which is 10 more than the “Call paths” value as seen in the QFlex Server configuration found at “Server Status > Edit” next to your physical QFlex server name. See graphic below with highlighted parameter:

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Control Access

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Parameters

Finally, associate this RTP configuration to a physical QFlex server and then download the configuration to the associated QFlex server.

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4.3 Server Configuration Access “Server Configuration” from the QFlex EMS menu. In the “Server Configuration Selection” window, click on the link which represents your physical QFlex SBC.

Ensure the following parameters are configured correctly for the IOT environment.

Add LAN-side Static Route

Since the PBX is on a different subnet in the LAN in this IOT lab environment, a LAN Static Route must be created. Click the “Add” button in the LAN Static Route window (see above graphic).

The following interface will be presented. Complete the three attributes as seen in Figure 15.

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Figure 15 - LAN Static Route Attributes

• “Address” is the IP-PBX IP network.

• “Netmask” is the IP-PBX IP network subnet mask.

• “Gateway” is the QFlex IP Gateway used by QFlex to reach the IP-PBX.

IP Address White List

Specify the IP Addresses from which calls are to be accepted by QFlex. Perform this exercise for LAN and WAN side sources, i.e., the LAN-side PBX and the WAN-side SBC.

On the LAN side, add an entry for each CUCM node that can send traffic to QFlex.

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And now for the WAN side:

Be sure to click the “Set Iptables White List” button to submit the configuration to the QFlex server.

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5 Cisco Unity Connection

Cisco Unity Connection provides the Voice Mail and Auto Attendant functionality to the enterprise eco system.

5.1 Create CUCM Integration The Cisco Unity Connection must be coupled to a CUCM. Do this by configuring a “Phone System” and “Port Group” as seen in the graphics below.

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5.2 Create a User A User account must be created on Unity to correspond to the CUCM extension. From the “Users > Users” menu, click the “Add New” button to create a new Unity User as seen in the graphics below.

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• Save the entry, then complete the remaining of the User record as seen below.

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• Synchronize the “Extension” value here with what is provisioned for the corresponding user in CUCM.

• Reference the “Phone System” which represents your CUCM from the value as defined for the CUCM in section 5.1.

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