Alcatel-Lucent OmniPCX

Enterprise SIP

I n t e g r a t i o n Te c h n i c a l N o t e
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Edition Note
This is the tenth edition of the Alcatel-Lucent OmniPCX Enterprise SIP Integration Technical
Note. Information in this document applies to CallXpress version 8.20.
P/N 1081-50553-00 Rev 10 October 2011
Produced by the AVST Technical Communications Department
Applied Voice & Speech Technologies, Inc., reserves the right to revise this guide without
notice.

Copyright Notice
2010 Applied Voice & Speech Technologies, Inc. (AVST). No part of this publication may
be reproduced, transmitted, transcribed, stored in a retrieval system, adapted, or translated
into any language in any form by any means without the written permission of AVST.
Trademarks, service marks, products names, company names or logos of AVST are
protected by trademark and other laws of the United States, as well as international
conventions and the laws of other countries. Other such properties that are not owned by
AVST may not be used without the express permission from their owners.

Disclaimer
AVST exercises care in researching other manufacturers specifications but disclaims all
responsibility for their accuracy. All specifications shown here are subject to change without
notice.

Trademarks
CallXpress and AVST are a registered trademarks and Applied Voice & Speech
Technologies, and the AVST logo is a trademark of Applied Voice & Speech Technologies,
Inc.
Windows is a registered trademark of Microsoft Corporation
RADVISION is a registered trademark of RADVISION Inc.
All other products, brand names, or companies mentioned in this document may be
trademarks or registered trademarks of their respective companies.

Applied Voice & Speech Technologies, Inc.

27042 Towne Centre Drive, Suite 200
Foothill Ranch, California 92610
(949) 699-2300
http://www.AVST.com

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Table of Contents
Overview ................................................................................................................................... 1
References ............................................................................................................................ 2
Features Supported by This Integration ................................................................................. 2

Critical Application Considerations ........................................................................................ 4

Installation Requirements ....................................................................................................... 5

Telephone System Requirements .......................................................................................... 5
CallXpress Requirements ...................................................................................................... 5

Programming the Telephone System ..................................................................................... 6

Verifying the Network Settings ............................................................................................... 6
Creating the SIP Trunk Group ............................................................................................... 7
Configuring the SIP Gateway ................................................................................................ 8
Configuring the CallXpress Server as a Trusted IP Address .................................................. 9
Configuring the SIP Registration Period ................................................................................ 9
Restarting the Call Server Processing Unit ...........................................................................10
Configuring the Voicemail Settings .......................................................................................10

Configuring CallXpress .......................................................................................................... 11

Configuring Switch Redundancy and Failover ......................................................................14
Completing the Integration ...................................................................................................17

Changing the Network Binding Order on the CallXpress Platform .....................................18

Windows Server 2003 ..........................................................................................................18
Windows Server 2008 R2 .....................................................................................................19

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Overview
This Integration Technical Note (ITN) is written for dealers who are experienced with CallXpress
and are familiar with its procedures and terminology. This document also assumes that you are
familiar with the features and programming of the Alcatel-Lucent telephone system.
This document describes how to integrate CallXpress with an Alcatel-Lucent telephone system,
using the Session Initiation Protocol (SIP) integration. This integration operates exclusively over
a TCP/IP-based network; it uses no analog or digital voice telephony ports, but passes voice
communication and signaling information over the network.

The Alcatel-Lucent integration is a SIP trunk integration, CallXpress registers itself as a single
voice mail pilot number at the telephone systems SIP proxy. The proxy routes all incoming calls
for CallXpress to that pilot number, and CallXpress assigns them to lines as needed. CallXpress
also sends outbound calls and MWI messages to the pilot number so that the proxy can act on
them.
This ITN documents the procedure for setting up the integration. The process consists of
programming the Alcatel-Lucent OmniPCX Enterprise and configuring CallXpress. Critical
application considerations are also documented.
Use this document in conjunction with the Installing CallXpress and Administering CallXpress
guides and with the CallXpress online help system.

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References
Information that is more current may be available from the following source:
World Wide WebAVST certified technicians can view or download documents and
program files from our web site: http://www.partnerxpress.com.
The following print documents are available for your reference:
Installing CallXpress
Administering CallXpress

Features Supported by This Integration

Tables 1 and 2 list the features supported using the Alcatel-Lucent OmniPCX Enterprise SIP
integration.
Table 1: Call forward to personal greeting support for these common call types
Divert to CallXpress on Supported
No Answer Yes
Busy Yes
Forward All Yes
Reject Call Yes

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Table 2: Alcatel-Lucent OmniPCX Enterprise SIP station set integration features supported
Feature Supported Notes
Automatic subscriber logon Yes
ANI/CLI Yes
Announce Busy greeting on forwarded calls Yes
Call screening Yes Note 1
Caller queuing Yes Note 2
DNIS Yes
End-to-end DTMF, attendant console Yes
End-to-end DTMF, proprietary telephones Yes
Fax Detection No
Internal calling party ID for reply Yes
Live record, integrated Yes
Live reply to sender Yes
Message notification callouts Yes
MWI, set/clear Yes
MWI, inband/outband Outband
Networking, analog Yes
Overflow from CallXpress to attendant Yes
Overflow to CallXpress from attendant Yes
PBX-provided disconnect signaling Yes
Revert to operator Yes
Transfers, blind Yes
Transfers, confirmed Yes
Transfers, fully supervised Yes
Transfers, monitored Yes
Trunk ID for call routing No
Notes
1. Only available when using supervised transfers
2. Caller Queuing is specific to each local Call Server. Call Servers within the system are
unaware of queued calls to the same subscriber on other Call Servers. For more
information, refer to the Critical Consideration Notes.

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Critical Application Considerations

Known limitations or conditions within the telephone system and CallXpress that affect the
integration performance are listed here. General recommendations are provided when ways to
avoid these limitations exist.
If your CallXpress server platform is a component of two or more local or wide area
networks (LANs or WANs), you must make sure that this integration does not interfere
with the normal network operation of the server. By default, CallXpress uses the primary
(public) network interface card (NIC) in the platform, the first NIC in the network binding
order. If you want CallXpress to use a NIC other than the first one, you must make
several required configuration changes. It is much easier to configure the Integration to
use another NIC by simply setting the integration parameter Local IP Address to bind
on to the address of the NIC card connected to the PBX. For more information, refer to
the section, Changing the Network Binding Order.
You must install the Windows Quality of Service (QoS) packet scheduler on the Call
Server platform if you plan to use the QoS feature of CallXpress when integrating with
the DMG. For more information about installing and configuring the QoS packet
scheduler, refer to Windows Help.
CallXpress supports G.729a with support for annex b on the incoming audio stream only.
CallXpress does not transmit annex b packets.
When codec negotiation takes place between CallXpress and the PBX, CallXpress
always offers the G.729a audio format as an option. You may configure G.729a as the
preferred codec in CallXpress; however, the decision whether to use G.729a is always
made by the PBX.
The Call Queuing feature does not transcend the Call Server. Calls may be queued on
multiple Call Servers for the same subscriber but Call Servers do not have knowledge of
calls in the queue on other Call Servers within the system. Callers may be prompted with
specific information about their place in the queue; however, the information pertains to
the specific Call Server on which their call is queued.
You must configure the Hunt Group Access Code in the Switch Section Options dialog
box and must match the Voice Mail Dir. No. configured in the Configure the Voicemail
Settings section.
On PBX versions prior to 9.0, the Call Server Processing Unit must be restarted after
you configure the SIP gateway.
When using Alcatel-Lucent OMNIPCX v9 or higher, the node name of the OXE must be
a fully qualified domain name (FQDN).

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Installation Requirements
Review the following information before performing any of the procedures in this document. To
successfully install this integration, you must meet the installation requirements for both the
telephone system and CallXpress.

Telephone System Requirements

One of the following releases:
Alcatel-Lucent OmniPCX Enterprise release 8.0, version g1.302, patch 6d
Alcatel-Lucent OmniPCX Enterprise release 9.0
Alcatel-Lucent OmniPCX Enterprise release 9.1
CPU Type: CPU80s1 and 128MB RAM
One SIP Gateway license for the system (#185) and one SIP Network Links license
(#188) for each CallXpress port

CallXpress Requirements
CallXpress version 8.20
AVST software key and feature file with the Alcatel OmniPCX Enterprise SIP integration
enabled and one RADVISION SIP and RTP license enabled for each port involved in
the integration
One or two network interface cards

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Programming the Telephone System

Follow the recommendations and programming examples in this section to program the
telephone system for integration with CallXpress. Programming examples show commands and
parameters that are necessary for integration; they do not represent PBX programming in its
entirety.
The installing technician should be familiar with programming the telephone system.
Programming is performed from the OmniPCX-programming terminal. For detailed
programming information on this software version or other OmniPCX software versions, refer to
the appropriate Alcatel OmniPCX Enterprise documentation.
All programming is performed through the Manager application. To use the Manager application,
access the main Call Server Processing Unit through a command line program such as Telnet.
Type your login and password, and then type mgr to enter the Manager application.

Verifying the Network Settings

To verify the network settings
1. From the main menu select Translator, and then press Enter.
2. Select Network Routing Table, and then press Enter.
3. Select Review/Modify, and then press Enter.
4. Press Enter to select All Instances, and then press Ctrl+V.
5. For the node you have chosen, verify the settings as shown in the example below.
Review/Modify: Network Routing Table
Node Number (reserved) : 1
Instance (reserved) : 1
Network Number : 10
Rank of First Digit to be Sent : 1
Incoming Identification Prefix : -----
Protocol Type + ABC-F
Numbering Plan Descriptor ID : 11
ARS Route List : 0
Schedule Number : -1
ATM Address ID : -1
Network Call Prefix : -----
City/Town Name : -----
Send City/Town Name : False
Associated Ext SIP Gateway : -1
Enable UTF8 Name Sending : True

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Creating the SIP Trunk Group

To create the SIP trunk group
1. From the main menu select Trunk Groups, and then press Enter.
2. Select Create, and then press Enter.
3. Change the settings as shown in the example below, and then press Ctrl+V to save your
changes.
Create: Trunk Groups
Node Number (reserved) : 1
Trunk Group ID : 10
Trunk Group Type + T2
Trunk Group Name : SIP Local
UTF-8 Trunk Group Name : -----
Number Compatible With : -1
Remote Network : 10
Shared Trunk Group + False
Special Services + Nothing

Create: Trunk Groups

Node Number : 1
Transcom Trunk Group + False
Auto.reserv.by Attendant + False
Overflow Trunk Group No. : -1
Tone on Seizure + False
Private Trunk Group + False
Q931 Signal Variant + ABC-F
SS7 Signal Variant + No variant
Number of Digits to Send : 0
Channel Selection Type + Quantified
Auto.DTMF Dialing on Outgoing Call + YES
T2 Specifications + SIP
Public Network COS : 31
DID Transcoding + False
Can Support UUS in SETUP + True

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Configuring the SIP Gateway

To configure the SIP gateway
1. From the main menu select SIP, and then press Enter.
2. Select SIP Gateway, and then press Enter.
3. Select Review/Modify, and then press Enter.
4. Press Enter to select All Instances, and then press Ctrl+V.
5. Change the settings as shown in the example below and then press Ctrl+V to save your
changes.
NOTE The machine name is the domain name for CallXpress. This value is case
sensitive and is unique for your site.

Review/Modify: SIP Gateway

Node Number (reserved) : 1
Instance (reserved) : 1
Instance (reserved) : 1
SIP Subnetwork : 10
SIP Trunk Group : 10
IP Address (FQDN) : 123.45.6.789
Machine Name Host : alcatel-oxe (See note
*)
SIP Proxy Port Number : 5060
SIP Subscriber Min Duration : 1800
SIP Subscriber Max Duration : 86400
Session Timer : 1800
Min. Session Timer : 900
Session Timer Method + RE_INVITE
DNS Local Domain Name : -----
DNS Type + DNS A
SIP DNS1 IP Address : -----
SIP DNS2 IP Address : -----
SDP IN 180 + True
Cac SIP-SIP + False
INFO method for remote extension + False
Dynamic Payload type for DTMF : 97

* NOTE Alcatel-Lucent OmniPCX Enterprise release 9.0 and later requires a fully
qualified domain name (FQDN) as the Machine Name-Host.

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Configuring the CallXpress Server as a Trusted IP Address

To configure the IP address
1. From the main menu select SIP, and then press Enter.
2. Select Trusted IP Addresses, and then press Enter.
3. Select Create, and then press Enter.
4. Type the IP address of the CallXpress server in the trusted address line, and then
press Ctrl+V to save your changes.
IMPORTANT You must restart the Call Server Processing Unit to apply your changes on
PBX versions prior to 9.0.

Configuring the SIP Registration Period

To configure the SIP Registration Period
1. From the main menu select SIP, and then press Enter.
2. Select SIP Registrar, and then press Enter.
3. Select Review/Modify, and then press Enter.
4. Press Enter to select All Instances, and then press Ctrl+V.
5. Change the SIP Min Expiration Date as shown in the example below, and then press
Ctrl+V to save your changes.
NOTE The default value of the SIP Min Expiration Date (expiry period) on the Alcatel
OMNI PCX is 3600 seconds. This value must be less than or equal to the value set in
the field, Registration period in seconds (default 90 seconds) in the CallXpress
Required Parameter Settings for the CallXpress fail-over to work effectively.

IMPORTANT You must restart the Call Server Processing Unit to apply your changes on
PBX versions prior to 9.0.

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Restarting the Call Server Processing Unit

On PBX software versions prior to 9.0 you must restart the Call Server Processing unit to apply
the programming changes.
To restart the Call Server Processing unit
1. From the main menu press Ctrl+C to quit the Manager application.
2. In the command line type killall sipmotor, and then press Enter.

Configuring the Voicemail Settings

NOTE This procedure programs the Messages button on telephones system-wide.
To configure the voicemail settings
1. From the main menu select Applications, and then press Enter.
2. Select External Voicemail, and then press Enter.
3. Select Create, and then press Enter.
4. Change the settings as shown in the example below, and then press Ctrl+V to save your
changes.
IMPORTANT Do not use 0000 as your SIP password. Instead, choose a unique
password.

Create: External Voice Mail

Node Number (reserved) : 1
Instance (reserved) : 1
Voice Mail Dir. No. : 5000
Directory Name : CallXpress
Connection COS : 0
Public Network COS : 2
Entity Number : 1
Cost Center ID : 255
Charging COS + Justified
URL UserName : -----
Remote SIP Domain + False
URL Domain : node000000
SIP Password : 0000
Confirm : 0000
Register On Line Number : -----
Register URL (Username) : -----
Register URL (Remote domain) + False
Register URL (Domain) : -----
Register Password : -----
Confirm : -----

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Configuring CallXpress
Once the telephone system is programmed, you must configure CallXpress for the integration.
During setup, you need the following information, specific to this integration, to configure
CallXpress.
To configure CallXpress
1. If you are configuring the Call Server for the first time, enter the name of your site in the
Site Name box of the Installation Configuration dialog box, the mailbox length in digits in
the Mailbox Length box, and the first extension number for the first line in the First
Extension box.
2. Select Alcatel-Lucent as the manufacturer, OmniPCX Enterprise as the model, and
then SIP as the integration type.
3. Click OK. The Switch Options dialog box displays.
4. Make any changes to the default settings your site requires, and then click OK.
5. The Integration Options dialog box, Required Parameters View displays.

6. Enter the information shown in Table 3.

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Table 3: Required Parameters Settings

Field Value
SIP Server Address Enter the TCP/IP address or an FQDN. For
example, the IP address 123.45.6.789 or Machine
Name-Host as configured on the Review/Modify
SIP Gateway screen. (See page 8 of this
document.) NOTE Alcatel-Lucent OmniPCX
Enterprise release 9.0 and later requires the
Machine Name to be a fully qualified domain
name (FQDN). When integrating with release 9.0
or later, use the Machine Name-Host in this field.
IMPORTANT This value must match the
configuration of the SIP Gateway.
SIP Server Port The port on which CallXpress listens for incoming
SIP messages. The default is 5060.
IMPORTANT This value must match the
configuration of the SIP proxy.
SIP Domain Name The domain name as configured on the switch.
IMPORTANT This value is case-sensitive and
must match the Machine NameHost configured
on the Review/Modify screen of the SIP Gateway.
For example, alcatel-oxe (See page 8 of this
document.)
Transport for outgoing SIP A default value of UDP is already selected, but
Messages you can select TCP, depending on your
configuration.
SIP Device Name The voice mail pilot number as configured on the
switch. This value matches the hunt group number
configured in the Alcatel Switch Section of
CallXpress.
IMPORTANT The SIP device name must match
the Voice Mail Dir. No. configured on the Create:
External Voice Mail screen. (See page 11 of this
document.)
PBX Password The password associated with the SIP Device
Name (for example, 0000).
IMPORTANT This password must match the SIP
password configured on the Create: External
Voice Mail screen. (page 11)
Registration period in seconds Controls the frequency of registration refreshes
and not only helps the switch to route calls to
voicemails but also enables CallXpress to detect
any network switch failure and initiate the failover
process. The smaller this value, the sooner the
failover initiates. The default is 90.
Local IP Address to bind on Select the local TCP/IP address of the CallXpress
machine. This is a drop-down box and displays all

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Field Value
available local TCP/IP addresses.
SIP Local Connection Port The port on which CallXpress listens for incoming
SIP messages. The default value is 5060.
SIP parser qualifier string The IP address of the Call Server platform. This
field allows the telephone system to locate the SIP
parser file on the platform.
Media packet size (milliseconds) CallXpress sends/receives packets containing the
number of milliseconds worth of audio data set
here. The default value is 20.
7. From the Local Integration Settings View, select Software DTMF Detection Settings.
The Software DTMF Settings view displays.
8. Confirm the DTMF Detection Type parameter is set to Hardware, the default value.
9. Click OK. The Switch Section Options dialog box displays.
10. In the Switch Section Options dialog box, select an Incoming Hunt Mode, and then type
the Hunt Group Access Code. In this example, 5000
NOTE This integration only supports terminal, circular, reverse terminal, and reverse
circular hunt modes.

11. Click Apply to save your changes.

12. In the AT_SysCfg information dialog box, click Yes to open the Board Options dialog
box.
13. In the Board Options dialog box, select RADVISION as the Manufacturer, SIP STACK
(RADVISION) as the Model, and then SIP IP RTP as the Protocol.
14. Click the Lines tab, and configure callouts for the application. For information on
configuring callout settings, see the topic Configuring Callout Settings, in the online
help system.
15. Click OK to save all changes.
The settings related to the telephone system on the Switches tab, and the other integration-
specific settings not mentioned in this section, are filled in correctly for a typical site when you
select the correct telephone system during setup. If you need to customize settings on the
Switch Sections or Integrations tab to meet requirements specific to your site, do so now.

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Configuring Switch Redundancy and Failover

Alcatel-Lucent OXE nodes can be configured for redundancy/failover in multiple ways.
The OXE nodes can be configured to share an IP address
An OXE node whose current role is MAIN, owns the IP address and communicates
with CallXpress and other devices through that IP address. If the roles between the OXE
nodes switch, the ownership of the IP address is also switched in a seamless fashion.
For such a configuration, the switch failover is transparent to CallXpress and it is not
necessary to configure CallXpress for failover.
The OXE nodes can be configured so that they share a common Fully Qualified Domain
Name (FQDN) but separate IP addresses as displayed in the following PBX
programming example of the DNS Delegation Properties dialog box.
NOTE The two OXE nodes may share a common domain name or SRV record instead
of a FQDN.

This is a typical configuration when the servers are geographically separated and are located on
different subnets (also known as spatial redundancy). The DNS server on the IP network is
responsible to resolve the FQDN to an IP address. The DNS server is configured to delegate
the request to the unique OXE nodes for the FQDN associated with the OXE. The OXE node
whose current role is MAIN responds to the DNS queries and provides its IP address. If the
OXE role switches, the node responding to the DNS queries also switches. Therefore, the
switch failover is transparent to CallXpress as long as it is configured to work with FQDN
instead of an IP address. The failover server configuration should be configured to use the
FQDN.
CallXpress can support switch failover even if the OXE nodes are not configured to use a
FQDN. In this case, follow the steps outlined below to configure the IP addresses of the primary
and failover OXE nodes.

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To add a failover server

This version of CallXpress enables failover server(s) configuration. You can add one or more
failover servers on the Integrations Options dialog box as follows:
1. From the System Configuration utility click the Integrations tab, highlight the Alcatel-
Lucent OmniPCX Enterprise SIP integration, and then click Edit.
2. From the View drop-down box on the Integration Options dialog box, select Failover
Server Settings. The Integration Options dialog box Failover Server Settings view
displays.

3. Click the Add Failover Server button. Two new rows are added to configure the
secondary SIP server.
4. In the Failover Server Settings view, enter Secondary SIP Server Address and the
Secondary SIP Server Port.

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Table 4: Secondary SIP Server Address and the Secondary SIP Server Port example
Field Value
(1) Secondary SIP Server Enter the TCP/IP address or an FQDN of
Address the secondary node. For example, the IP
address 123.45.6.789 as displayed on the
Review/Modify SIP Gateway screen. (See
page 8 of this document.)
NOTE Alcatel-Lucent OmniPCX Enterprise
release 9.0 requires the Machine Name to
be a fully qualified domain name. Therefore,
when integrating with release 9.0 or later,
use the Machine name field as displayed on
the Review/Modify SIP Gateway screen.
The default value is blank.
IMPORTANT This value must match the
configuration on the Gateway of the
secondary node.
(1) Secondary SIP Server Port The default value is 5060
5. From the View drop-down box, select Integration Specific Parameters. The Integration
Options dialog box Integration Specific Parameters view displays.

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6. Enter the information as shown in Table 5.

Table 5: Integration Specific Parameters

Field Value
Enable SIP server failover Select this check box to allow for failover and to
enable the failover server setting changes.
Delay (in ms) between Failover The delay in milliseconds before CallXpress attempts
attempts to register its port with the SIP server. The default is
1000 ms
Incoming off hook delay 800
Outgoing off hook delay 0
On hook delay 300
Media streaming port number 48004
Enable IP QoS Select the check box to enable
Token Rate (bytes/second) 12000
DSCP value for Media 46
7. Click Apply, and then click OK to close the Integration Options dialog box.

8. To add another failover server repeats steps 1-7.

To remove a Failover Server

1. From the System Configuration utility click the Integrations tab, highlight the Alcatel-
Lucent OmniPCX Enterprise SIP integration, and then click Edit.
2. From the View drop-down box on the Integrations Options dialog box, select Failover
Server Settings. The Integration Options dialog box Failover Server Settings view
displays.
3. Click the Remove Failover Server button.
4. At the confirmation prompt, click Yes to confirm the deletion. If multiple servers are
listed, the last one on the list is deleted first.
5. Click Apply, and then click OK to close the Integration Options dialog box.

Completing the Integration

Now you are ready to finish installing CallXpress. See the guides Installing CallXpress and
Administering CallXpress, or refer to the CallXpress online help system, for instructions. For
general information on integrations, you may also wish to consult Integrating CallXpress with
the Telephone System, in the guide Installing CallXpress, and the topic Integrate the
Telephony Server with the telephone system, in the online help system.

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Changing the Network Binding Order on the

CallXpress Platform
If your CallXpress server platform is a component of two or more local or wide area networks
(LANs or WANs), you must make sure that this integration does not interfere with the normal
network operation of the server. By default, CallXpress uses the primary (public) network
interface card (NIC) in the platform, the first NIC in the network binding order. If you want
CallXpress to use a NIC other than the first one, you must make several required configuration
changes. It is much easier to configure the Integration to use another NIC by simply setting the
integration parameter Local IP Address to bind on to the address of the NIC card connected to
the PBX.
NOTE The operating system gives precedence to the first network connection in the list followed
by the remaining connections based on their position in the list.

The instructions in this section ensure that the binding order is correct when you set up the
integration. However, if you replace a NIC on the CallXpress server platform later, the platform's
operating system registers the new adapter at the bottom of its binding order. Restoring the
original binding order should correct any problems caused by the change.
IMPORTANT The following procedure shifts the binding order of the network interface cards. To
determine which NIC is associated with a specific network connection, right-click the connection
in the Network Connections window, and then select Properties.

Windows Server 2003

To change the binding order of multiple NICs
1. From the taskbar, select Start | Settings | Control Panel.
2. In the Control Panel, double-click Network Connections (or Network and Dial-Up
Connections).
3. From the menu bar on the Network Connections window select Advanced, and then
click Advanced Settings.
4. On the Adapters and Bindings tab of Advanced Settings, click the network connection
that serves CallXpress.
5. Click the up arrow button to the right of the Connections list as many times as needed to
move the connection to the top of the list.
6. Click OK, and then close the Network Connections window and the Control Panel.
7. Shut down and restart the CallXpress server platform.

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I n t e g r a t i o n Te c h n i c a l N o t e 19

Windows Server 2008 R2

To change the binding order of multiple NICs
1. From the taskbar, select Start | Settings | Control Panel.
2. In the Control Panel, double-click Network and Sharing Center (use small icons).
3. On the left pane, select Change Adapter Settings.
4. Press Alt to display the menu bar.
5. On the menu bar, select Advanced, and then click Advanced Settings.
6. On the Adapters and Bindings tab of Advanced Settings, click the network connection
that serves CallXpress.
7. Click the up arrow button to the right of the Connections list as many times as needed to
move the connection to the top of the list.
8. Click OK, and then close the Network Connections window and the Control Panel.

1081-50553-00 Rev 10, v8.20 October 2011