Application Notes for Configuring Cincinnati Bell eVantage IP

Application Notes for Configuring Cincinnati Bell eVantage IP
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Cincinnati Bell eVantage
IP Service with Avaya Communication Server 1000E 7.5,
Avaya Aura® Session Manager 6.2, Avaya Session Border
Controller for Enterprise 4.0.5 – Issue 1.0
Abstract
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between Cincinnati Bell eVantage IP Service and an Avaya SIP-enabled enterprise
solution. The Avaya solution consists of Avaya Communication Server 1000E, Avaya Aura®
Session Manager, Avaya Session Border Controller for Enterprise and various Avaya
endpoints.
Cincinnati Bell is a member of the Avaya DevConnect Service Provider program. Information
in these Application Notes has been obtained through DevConnect compliance testing and
additional technical discussions. Testing was conducted via the DevConnect Program at the
Avaya Solution and Interoperability Test Lab.
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Table of Contents
1.
2.
Introduction ............................................................................................................................. 4
General Test Approach and Test Results ................................................................................ 4
2.1. Interoperability Compliance Testing ................................................................................ 4
2.2. Test Results ...................................................................................................................... 5
2.3. Support ............................................................................................................................. 6
3. Reference Configuration ......................................................................................................... 6
4. Equipment and Software Validated ........................................................................................ 7
5. Configure Avaya Communication Server 1000E ................................................................... 8
5.1. Administer an IP Telephony Node ................................................................................... 9
5.1.1. Obtain Node IP Address ........................................................................................... 9
5.1.2. Terminal Proxy Server (TPS) ................................................................................. 11
5.1.3. Quality of Service (QoS) ........................................................................................ 12
5.1.4. Voice Gateway and Codecs .................................................................................... 13
5.1.5. SIP Gateway............................................................................................................ 14
5.1.6. Synchronize Node Configuration ........................................................................... 17
5.2. Virtual Superloops.......................................................................................................... 19
5.3. Media Gateway .............................................................................................................. 19
5.4. Virtual D-Channel, Routes and Trunks .......................................................................... 23
5.4.1. Virtual D-Channel Configuration ........................................................................... 23
5.4.2. Routes and Trunks Configuration ........................................................................... 25
5.5. Dialing and Numbering Plans ........................................................................................ 27
5.5.1. Route List Block ..................................................................................................... 27
5.5.2. NARS Access Code ................................................................................................ 29
5.5.3. Numbering Plan Area Codes .................................................................................. 30
5.5.4. Special Numbers to Route to Session Manager ...................................................... 32
5.5.5. Incoming Digit Translation ..................................................................................... 33
5.6. Zones and Bandwidth ..................................................................................................... 34
5.7. Example CS1000E Telephone Users ............................................................................. 36
5.7.1. Example SIP Phone DN 7108, Codec Considerations............................................ 36
5.7.2. Example Digital Phone DN 7107 with Call Waiting .............................................. 37
5.7.3. Example Analog Port with DN 7106, Fax .............................................................. 38
5.8. Save Configuration ......................................................................................................... 39
6. Configure Avaya Aura® Session Manager .......................................................................... 40
6.1. Avaya Aura® System Manager Login and Navigation ................................................. 40
6.2. Specify SIP Domain ....................................................................................................... 41
6.3. Add Location .................................................................................................................. 42
6.4. Adaptations..................................................................................................................... 45
6.5. Add SIP Entities ............................................................................................................. 48
6.6. Add Entity Links ............................................................................................................ 52
6.7. Add Routing Policies ..................................................................................................... 53
6.8. Add Dial Patterns ........................................................................................................... 54
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6.9. Add/Verify Avaya Aura® Session Manager Instance ................................................... 57
7. Configure Avaya Session Border Controller for Enterprise ................................................. 59
7.1. Network Management .................................................................................................... 62
7.2. Routing Profile ............................................................................................................... 63
7.3. Topology Hiding Profile ................................................................................................ 65
7.4. Server Interworking Profile............................................................................................ 67
7.4.1. Server Interworking Profile – Enterprise ................................................................ 67
7.4.2. Server Interworking Profile – Cincinnati Bell ........................................................ 71
7.5. Server Configuration ...................................................................................................... 74
7.5.1. Server Configuration – Session Manager ............................................................... 74
7.5.2. Server Configuration - Cincinnati Bell ................................................................... 77
7.6. Media Rule ..................................................................................................................... 81
7.7. Signaling Rule ................................................................................................................ 83
7.8. Application Rule ............................................................................................................ 85
7.9. Endpoint Policy Group ................................................................................................... 86
7.10.
Media Interface ........................................................................................................... 88
7.11.
Signaling Interface ...................................................................................................... 89
7.12.
End Point Flows - Server Flow................................................................................... 89
8. Cincinnati Bell eVantage IP Service Configuration ............................................................. 93
9. Verification ........................................................................................................................... 93
9.1. Avaya Communication Server 1000E Verification ....................................................... 93
9.1.1. IP Network Maintenance and Reports Commands ................................................. 93
9.1.2. System Maintenance Commands ............................................................................ 95
9.2. Avaya Aura® Session Manager Verification ................................................................. 97
9.3. Avaya Session Border Controller for Enterprise Verification ....................................... 97
9.3.1. Incidents .................................................................................................................. 98
9.3.2. Diagnostics .............................................................................................................. 99
9.3.3. Trace Settings........................................................................................................ 100
10. Conclusion ....................................................................................................................... 102
11. Additional References ...................................................................................................... 102
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1. Introduction
These Application Notes describe a sample configuration of Avaya Communication Server
1000E release 7.5 Avaya Aura® Session Manager 6.2, and Avaya Session Border Controller for
Enterprise 4.0.5 (Avaya SBCE) integration with Cincinnati Bell eVantage IP Service.
In the sample configuration, the Avaya Session Border Controller for Enterprise is used as an
edge device between Avaya Customer Premise Equipment (CPE) and Cincinnati Bell eVantage
IP Service. The Avaya SBCE performs SIP header manipulation and provides Network Address
Translation (NAT) functionality to convert the private Avaya CPE IP addressing to IP addressing
appropriate for the Cincinnati Bell eVantage IP Service access method.
The Cincinnati Bell eVantage IP Service solution is a turn-key business trunking solution for
customers. Cincinnati Bell eVantage IP Service provides customers with a single IP connection
that converges voice and data services to drive optimization, reduce costs, and offer enhanced
features not typically available in the traditional PSTN network. Voice services, such as local,
long distance and toll free calling, as well a high speed data and Internet services, are the primary
applications of the Cincinnati Bell eVantage solution.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.
2. General Test Approach and Test Results
The general test approach was to configure a simulated enterprise site using Avaya
Communication Server 1000E (CS1000E), Session Manager, and Avaya SBCE to connect to the
public Internet using a broadband connection. The enterprise site was configured to connect to
Cincinnati Bell eVantage IP Service. This configuration (shown in Figure 1) was used to
exercise the features and functionality listed in Section 2.1.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.
2.1. Interoperability Compliance Testing
To verify SIP trunking interoperability, the following features and functionality were covered
during the interoperability compliance test:
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Incoming PSTN calls to various phone types. Phone types included UNIStim, SIP,
digital, and analog telephones at the enterprise. All inbound PSTN calls were routed to
the enterprise across the SIP trunk from the service provider
Outgoing PSTN calls from various phone types. Phone types included UNIStim, SIP,
digital, and analog telephones at the enterprise. All outbound PSTN calls were routed
from the enterprise across the SIP trunk to the service provider
Inbound and outbound PSTN calls to/from Avaya one-X Communicator (soft client)
Various call types including: local, long distance, and outbound toll-free
Codecs G.729A, G.729B and G.711MU
DTMF transmission using RFC 2833
G711 Fax
Caller ID presentation and Caller ID restriction
Voicemail navigation for inbound and outbound calls
User features such as hold and resume, transfer, and conference
Items not supported or not tested included the following:
Inbound toll-free, operator, operator services (0 + 10 digits) and emergency calls (911)
are supported but were not tested as part of the compliance test
Calls forwarded off-net were not supported on the test circuit used for the compliance
test, but Cincinnati Bell eVantage IP Service production environment does support these
types of calls.
SIP REFER method is not supported by Avaya CS1000E
CS1000E Mobile-X features were not tested
2.2. Test Results
Interoperability testing of Cincinnati Bell eVantage IP Service was completed with successful
results for all test cases with the exception of the observations/limitations described below.
Calling Party Number (PSTN transfers): The calling party number displayed on the
PSTN phone is not updated to reflect the true connected party on calls that are
transferred to the PSTN. After the call transfer is complete, the calling party number
displays the number of the transferring party and not the actual connected party. The
PSTN phone display is ultimately controlled by the PSTN provider, thus this behavior is
not necessarily indicative of a limitation of the combined Avaya/Cincinnati Bell
eVantage IP Service solution. It is listed here simply as an observation.
T.38 Fax: At the time of original publication of these Application Notes, Cincinnati Bell
eVantage IP Service supported fax over T.38 within their local calling area only. Any
fax calls placed outside of the Cincinnati Bell local calling area will be transferred using
G.711 codec. The recommended workaround is to configure the CS1000E fax endpoints
to use the G.711codec for outbound calling. See Section 5.7.3
Cincinnati Bell eVantage IP Service passed compliance testing.
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2.3. Support
For technical support on the Cincinnati Bell eVantage IP Service, contact Cincinnati Bell using
the Customer Care links at www.Cincinnati Bell.com.
3. Reference Configuration
Figure 1 illustrates the sample configuration used for the DevConnect compliance testing. The
configuration is comprised of the Avaya CPE location connected via an Internet connection to
the Cincinnati Bell eVantage IP Services. The Avaya CPE location simulates a customer site. At
the edge of the Avaya CPE location, an Avaya SBCE provides NAT functionality and SIP
header manipulation. The Avaya SBCE receives traffic from Cincinnati Bell eVantage IP
Service on port 5060 and sends traffic to the Cincinnati Bell eVantage IP Service using
destination port 5060, using the UDP protocol. For security reasons, any actual public IP
addresses used in the configuration have been replaced with private IP addresses. Similarly, any
references to real routable PSTN numbers have also been changed to numbers that cannot be
routed by the PSTN.
Figure 1: Avaya Interoperability Test Lab Configuration
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4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Avaya IP Telephony Solution Components
Component
Release
Avaya Communication Server 1000E running
Call Server: 7.50 .17 GA (CoRes)
on CP+DC server as co-resident configuration
Service Pack: 7.50.17_20120919
SSG Server: 7.50.17 GA
SLG Server: 7.50.17 GA
Communication Server 1000E Media
CSP Version: MGCC CD03
Gateway
MSP Version: MGCM AB02
APP Version: MGCA BA15
FPGA Version: MGCF AA19
BOOT Version: MGCB BA15
DSP1 Version: DSP4 AB06
BCSP Version: MGCC CD01
Avaya Session Border Controller for
4.0.5Q18
Enterprise
Avaya 1165E (UNIStim)
0626C8A
Avaya 1140E (SIP)
04.03.12.00
Avaya one-X Communicator (SIP)
CS6.1.1.02 SP1 36207
Avaya M3904 (Digital)
n/a
Avaya 6210 Analog Telephone
n/a
Cincinnati Bell eVantage IP Service Components
Component
Release
BroadSoft
Version 17
Table 1: Equipment and Software Tested
The specific configuration above was used for the compatibility testing.
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5. Configure Avaya Communication Server 1000E
This section describes the Avaya Communication Server 1000E configuration, focusing on the
routing of calls to Cincinnati Bell over a SIP trunk. In the sample configuration, Avaya
Communication Server 1000E Release 7.5 was deployed as a co-resident system with the SIP
Signaling Server, and Call Server applications all running on the same CP+DC server platform.
This section focuses on the SIP Trunking configuration. Although sample screens are illustrated
to document the overall configuration, it is assumed that the basic configuration of the Call
Server and SIP Signaling Server applications has been completed, and that the Avaya
Communication Server 1000E is configured to support analog, digital, UNIStim, and SIP
telephones. For references on how to administer these functions of Avaya Communication
Server 1000E, see Section 11.
Configuration will be shown using the web based Avaya Unified Communications Management
GUI. The Avaya Unified Communications Management GUI may be launched directly via
https://<ipaddress> where the relevant <ipaddress> in the sample configuration is 10.80.140.102.
The following screen shows an abridged log in screen. Log in with appropriate credentials.
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The Avaya Unified Communications Management Elements page will be used for configuration.
Click on the Element Name corresponding to CS1000 in the Element Type column. In the
abridged screen below, the user would click on the Element Name EM on cs1k-cpdc.
5.1. Administer an IP Telephony Node
This section describes how to configure an IP Telephony Node on the Communication Server
1000E.
5.1.1. Obtain Node IP Address
Expand System  IP Network on the left panel and select Nodes: Servers, Media Cards.
The IP Telephony Nodes page is displayed as shown below. Click <Node id> in the Node ID
column to view details of the node. In the sample configuration, Node ID 1005 was used.
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The Node Details screen is displayed with additional details as shown below. Under the Node
Details heading at the top of the screen, make a note of the TLAN Node IPV4 address. In the
sample screen below, the Node IPV4 address is 10.80.140.103. This IP address will be needed
when configuring Session Manager with a SIP Entity for the CS1000E in Section 6.5.
The following screen shows the Associated Signaling Servers & Cards heading at the bottom
of the screen, simply to document the configuration.
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5.1.2. Terminal Proxy Server (TPS)
On the Node Details screen, scroll down in the top window and select the Terminal Proxy
Server (TPS) link as show below.
Check the UNIStim Line Terminal Proxy Server check box and then click the Save button
(not shown).
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5.1.3. Quality of Service (QoS)
On the Node Details screen, scroll down in the top window and select the Quality of Service
(QoS) link as shown below.
Set the Control packets and Voice packets values to the desired Diffserv settings required on
the internal network. The default Diffserv values are shown below. Click on the Save button.
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5.1.4. Voice Gateway and Codecs
On the Node Details screen, scroll down in the top window and select the Voice Gateway
(VGW) and Codecs link as shown below.
The following screen shows the General parameters used in the sample configuration.
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Use the scroll bar on the right to find the area with heading Voice Codecs. Note that Codec
G.711 is enabled by default. The following screen shows the G.711 parameters used in the
sample configuration.
For the Codec G.729, ensure that the Enabled box is checked, and the Voice Activity Detection
(VAD) box is un-checked. In the sample configuration, the CS1000E was configured to include
G.729A and G.711 in SDP Offers, in that order. During compliance testing, the G.729B codec
was also tested by checking the Voice Activity Detection (VAD) box.
5.1.5. SIP Gateway
The SIP Gateway is the SIP trunk between the CS1000E and Session Manager. On the Node
Details screen, scroll down in the top window and select the Gateway (SIPGw) link as show
below.
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On the Node ID: <id> – Virtual Trunk Gateway Configuration Details page, enter the
following values and use default values for remaining fields.
Sip domain name:
Local SIP port:
Gateway endpoint name:
Application node ID:
Enter the appropriate SIP domain for the customer network.
In the sample configuration, avayalab.com was used in the
Avaya Solutions and Interoperability Test lab environment.
Enter 5060.
Enter a descriptive name.
Enter <Node id>. In the sample configuration, Node 1005
was used matching the node show in Section 5.1.1.
The values defined for the sample configuration are shown below.
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Scroll down to the SIP Gateway Settings  Proxy or Redirect Server: section.
Under Proxy Server Route 1, enter the following and use default values for remaining fields.
Primary TLAN IP address: Enter the IP address of the Session Manager SIP signaling
interface. In the sample configuration 10.64.19.210 was
used.
Port:
Enter 5060
Transport protocol:
Select TCP
The values defined for the sample configuration are shown below.
Scroll down and repeat these steps for the Proxy Server Route 2.
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Scroll down to the SIP URI Map section. The values defined for the sample configuration are
shown below. The Avaya CS1000E will put the “string” entered in the SIP URI Map in the
“phone-context=<string>” parameter in SIP headers such as the To and From headers. If the
value is configured to blank, the CS1000E will omit the “phone-context=” in the SIP header
altogether.
Scroll to the bottom of the page and click Save (not shown) to save SIP Gateway configuration
settings. This will return the interface to the Node Details screen.
5.1.6. Synchronize Node Configuration
On the Node Details screen click Save as shown below.
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Select Transfer Now on the Node Saved page as show below.
Once the transfer is complete, the Synchronize Configurations Files (NODE ID <id>) page is
displayed. Place a check mark next to the appropriate Hostname and click Start Sync. The
screen will automatically refresh until the synchronization is finished.
The Synchronization Status field will update from Sync required (as shown above) to
Synchronized (as shown below). After synchronization completes, place a check mark next to
the appropriate Hostname and click Restart Applications.
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5.2. Virtual Superloops
Expand System  Core Equipments on the left panel and select Superloops. In the sample
configuration, Superloop 4 is for the Media Gateway and Superloop 252 is the virtual Superloop
used by the IP phones and SIP trunks.
5.3. Media Gateway
Expand System  IP Network on the left panel and select Media Gateways. Click the link in
the Type column for the appropriate Media Gateway to be modified as shown below.
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The IPMG 4 0 Media Gateway Survivable (MGS) Configuration window appears. The
Telephony LAN (TLAN) IP Address under the DSP Daughterboard 1 heading will be the IP
Address in the SDP portion of SIP messages, for calls requiring a gateway resource. For
example, for a call from a digital telephone to the PSTN via Cincinnati Bell eVantage IP Service,
the IP Address in the SDP in the INVITE message will be 10.80.140.104 in the sample
configuration.
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Scroll down to the area of the screen containing VGW and IP phone codec profile and expand
it. The fax T.38 settings used for compliance testing is shown below.
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The Codec G.711 is enabled by default. Ensure that the Select box is checked for Codec G729A
and the VAD (Voice Activity Detection) box is un-checked. The Voice payload size of 20 can
be used with Cincinnati Bell eVantage IP Service for both G.729A and G.711. Click Save (not
shown) at the bottom of the window. Then click OK in the dialog box (not shown) to save the
IPMG configuration. During compliance testing, the G.729B codec was also tested by checking
the Voice Activity Detection (VAD) box. Scroll down and click Save and then click OK on the
new dialog box that appears to save the configuration.
After the configuration is saved, the Media Gateways page is displayed. Select the appropriate
Media Gateway and click Reboot to load the new configuration.
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5.4. Virtual D-Channel, Routes and Trunks
Avaya Communication Server 1000E Call Server utilizes a virtual D-channel and associated
Route and Trunks to communicate with the Signaling Server.
5.4.1. Virtual D-Channel Configuration
Expand Routes and Trunks on the left panel and select D-Channels. In the sample
configuration, there is a virtual D-Channel 15 associated with the Signaling Server.
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Select Edit to verify the configuration, as shown below. Verify DCIP has been selected for D
Channel Card Type field and the Interface type for D-Channel is set to Meridian Meridian
1(SL1). Under the Basic Options section, verify 128 is selected for the Output request Buffers
value.
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5.4.2. Routes and Trunks Configuration
In addition to configuring a virtual D-channel, a Route and associated Trunks must be
configured. Expand Routes and Trunks on the left panel and expand the customer number. In
the example screen that follows, it can be observed that Route 15 has 32 trunks in the sample
configuration.
Select Edit to verify the configuration, as shown below. As can be observed in the Incoming
and outgoing trunk (ICOG) parameter, incoming and outgoing calls are allowed. The Access
code for the trunk route (ACOD) will in general not be dialed, but the number that appears in
this field may be observed on Avaya CS1000E display phones if an incoming call on the trunk is
anonymous or marked for privacy.
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Further down in the Basic Configuration section verify the Node ID of signaling server of this
route (NODE) matches the node shown in Section 5.1.1. Also verify SIP (SIP) has been
selected for Protocol ID for the route (PCID) field. The Zone for codec selection and
bandwidth management (ZONE) parameter can be used to associate the route with a zone for
configuration of the audio codec preferences sent via the Session Description Protocol (SDP) in
SIP messaging. The D channel number (DCH) field must match the D-Channel number shown
in Section 5.4.1.
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Scroll down and expand the Basic Route Options section. Check the North American toll
scheme (NATL) and Incoming DID digit conversion on this route (IDC), input DCNO 0 for
both Day IDC Tree Number and Night IDC Tree Number as shown below. The DCNO is
created later on in Section 5.5.5.
5.5. Dialing and Numbering Plans
This section provides the configuration of the routing used in the sample configuration for
routing calls over the SIP Trunk between Avaya Communication Server 1000E and Session
Manager for calls destined for the Cincinnati Bell eVantage IP Service. The routing defined in
this section is simply an example and not intended to be prescriptive. Other routing policies may
be appropriate for different customer networks.
5.5.1. Route List Block
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Select Route List Block (RLB) on the Electronic Switched Network (ESN) page as
shown on the following page.
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The Route List Blocks screen is displayed. Enter an available route list index number in the
Please enter a route list index field and click to Add, or edit an existing entry by clicking the
corresponding Edit button. In the sample configuration, route list block index 15 is used. If
adding the route list index anew, scroll down to the Options area of the screen. If editing an
existing route list block index, select the Edit button next to the appropriate Data Entry Index as
shown below, and scroll down to the Options area of the screen.
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Under the Options section, select <Route id> in the Route Number field. In the sample
configuration route number 15 was used. Default values may be retained for remaining fields.
5.5.2. NARS Access Code
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Select ESN Access Codes and Parameters (ESN). Although not repeated below, this
link can be observed in the first screen in Section 5.5.1. In the NARS/BARS Access Code 1
field, enter the number the user will dial before the target PSTN number. In the sample
configuration, the single digit 9 was used.
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5.5.3. Numbering Plan Area Codes
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Scroll down and select Numbering Plan Area Code (NPA) under the appropriate
access code heading. In the sample configuration, this is Access Code 1, as shown below.
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Add a new NPA by entering it in the Please enter an area code box and click to Add or click
Edit to view or change an NPA that has been previously configured. In the screen below, it can
be observed that various dial strings such as 1303 and 1800 are configured.
In the screen below, the entry for 1303 is displayed. In the Route List Index, 15 is selected to use
the route list associated with the SIP Trunk to Session Manager as shown in Section 5.4.2.
Default parameters may be retained for other parameters. Repeat this procedure for the dial
strings associated with other numbering plan area codes that should route to the SIP Trunk to
Session Manager.
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5.5.4. Special Numbers to Route to Session Manager
In the testing associated with these Application Notes, special service numbers such as x11,
international calls, and operator assisted calls were also routed to Session Manager and
ultimately to the Cincinnati Bell eVantage IP Service. Although not intended to be prescriptive,
one approach to such routing is summarized in this section.
Expand Dialing and Numbering Plans on the left panel and select Electronic Switched
Network. Scroll down and select Special Number (SPN) under the appropriate access code
heading (as can be observed in the first screen in Section 5.5.3).
Add a new number by entering it in the Please enter a Special Number box and click to Add or
click Edit to view or change a special number that has been previously configured. In the screen
below, it can be observed that various dial strings such as 0, 011, 411 and 911 calls are listed.
Route list index 15 has been selected in the same manner as shown for the NPAs in the prior
section.
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5.5.5. Incoming Digit Translation
In general, the incoming digit translation can be used to manipulate the digits received for an
incoming call if necessary. Since Session Manager is present, Session Manager can be used to
perform digit conversion using an Adaptation as shown in Section 6.4, and digit manipulation
via the CS1000E Incoming Digit Translation table may not be necessary. If the DID number sent
by Cincinnati Bell is unchanged by Session Manager, then the DID number can be mapped to an
extension using the Incoming Digit Translation. Both Session Manager digit conversion and
CS1000E incoming digit translation methods were tested successfully.
Expand Dialing and Numbering Plans on the left panel and select Incoming Digit
Translation. Click on the Edit IDC button as shown below.
Click on the New DCNO to create the digit translation mechanism or if editing an existing one,
select the Edit DCNO button next to the appropriate Digit Conversion Tree Number. In this
example, Digit Conversion Tree Number (DCNO) 0 has been created as shown below.
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Detail configuration of the DCNO is shown below. The Incoming Digits can be added to map to
the Converted Digits which would be the CS1000E system phones DN. This DCNO has been
assigned to route 15 as shown in Section 5.4.2.
In the following configuration, the incoming DID 5135555180 will be translated to CS1000E
DN 2900.
5.6. Zones and Bandwidth
Zone configuration can be used to control codec selection and for bandwidth management. To
configure, expand System  IP Network on the left panel and select Zones as shown below.
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Select Bandwidth Zones. In the sample lab configuration, two zones are configured. In
production environments, it is likely that more zones will be required. Select the zone associated
with the virtual trunk to Session Manager and click Edit as shown below. In the sample
configuration, this is Zone number 99.
In the resultant screen shown below, select Zone Basic Property and Bandwidth
Management.
The following screen shows the Zone 99 configuration. Note that Best Bandwidth (BB) is
selected for the zone strategy parameters so that codec G.729A is preferred over codec G.711MU
for calls with Cincinnati Bell eVantage IP Service.
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5.7. Example CS1000E Telephone Users
This section is not intended to be prescriptive, but simply illustrates a sampling of the telephone
users in the sample configuration.
5.7.1. Example SIP Phone DN 7108, Codec Considerations
The following screen shows basic information for a SIP phone in the configuration. The
telephone is configured as Directory Number 7108. Note that the telephone is in Zone 1 and is
associated with Node 1005 (see Section 5.1). A call between this telephone and another
telephone in Zone 1 will use a best quality strategy (see Section 5.6) and therefore can use
G.711MU. If this same telephone calls out to the PSTN via the Cincinnati Bell eVantage IP
Service, the call would use a best bandwidth strategy, and the call would use G.729A.
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5.7.2. Example Digital Phone DN 7107 with Call Waiting
The following screen shows basic information for a digital phone in the configuration. The
telephone is configured as Directory Number 7107.
The following screen shows basic key information for the telephone. It can be observed that the
telephone can support call waiting with tone. Although not shown in detail below, to use call
waiting with tone, assign a key CWT – Call Waiting, set the feature SWA – Call waiting from
a Station to Allowed, and set the feature WTA – Warning Tone to Allowed.
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5.7.3. Example Analog Port with DN 7106, Fax
The following screen shows basic information for an analog port in the configuration that may be
used with a telephone or fax machine. The port is configured as Directory Number 7106.
When an analog port is used for a fax machine, Modem Pass Through Allowed (MPTA) can be
set to cause G.711 to be used instead of T.38 for fax calls, even if the zone configuration would
otherwise have resulted in G.729. For example, if MPTA is configured, and an inbound call
arrives from Cincinnati Bell eVantage IP Service, the CS1000E will respond with a 200 OK,
selecting G.711 for the call in the SDP answer, even if the SDP offer from Cincinnati Bell listed
G.729 before G.711. Similarly, for an outbound call with MPTA configured, the CS1000E will
send the INVITE with an SDP offer for G.711. See Section 2.2 for T.38 limitations with the
Cincinnati Bell eVantage IP Service.
To configure MPTA, scroll down to the Features area and locate the feature with description
Modem Pass Through. From the drop-down menu, select MPTA as shown below.
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5.8. Save Configuration
Expand Tools  Backup and Restore on the left panel and select Call Server. Select Backup
(not shown) and click Submit to save configuration changes as shown below.
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6. Configure Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. The procedures include
adding the following items:
SIP domain
Logical/physical Location that can be occupied by SIP Entities
SIP Entities corresponding to CS1000E, Avaya SBCE and Session Manager
Entity Links, which define the SIP trunk parameters used by Session Manager when routing
calls to/from SIP Entities
Routing Policies, which control call routing between the SIP Entities
Dial Patterns, which govern to which SIP Entity a call is routed
Session Manager Instance, corresponding to the Session Manager server to be administered
in System Manager.
It may not be necessary to create all the items above when creating a connection to the service
provider since some of these items would have already been defined as part of the initial Session
Manager installation. This includes items such as certain SIP domains, locations, SIP entities,
and Session Manager itself. However, each item should be reviewed to verify the configuration.
6.1. Avaya Aura® System Manager Login and Navigation
Session Manager configuration is accomplished by accessing the browser-based GUI of System
Manager, using the URL https://<ip-address>/SMGR, where <ip-address> is the IP address of
System Manager. Log in with the appropriate credentials and click on Log On (not shown). The
screen shown below is then displayed.
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Most of the configuration items are performed in the Routing Element. Click on Routing in the
Elements column shown above to bring up the Introduction to Network Routing Policy screen.
6.2. Specify SIP Domain
Create a SIP domain for each domain for which Session Manager will need to be aware in order
to route calls. For the compliance test, this includes the enterprise domain (avayalab.com).
Navigate to Routing  Domains and click the New button in the right pane (not shown). In the
new right pane that appears, fill in the following:
Name:
Type:
Notes:
Enter the domain name.
Select sip from the pull-down menu.
Add a brief description (optional).
Click Commit. The screen below shows the entry for the avayalab.com domain.
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6.3. Add Location
Locations can be used to identify logical and/or physical locations where SIP Entities reside for
purposes of bandwidth management and call admission control. To add a location, navigate to
Routing Locations in the left-hand navigation pane and click the New button in the right pane
(not shown).
In the General section, enter the following values. Use default values for all remaining fields:
Name: Enter a descriptive name for the location.
Notes: Add a brief description (optional).
The Location Pattern was not populated. The Location Pattern is used to identify call routing
based on IP address. Session Manager matches the IP address against the patterns defined in this
section. If a call is from a SIP Entity that does not match the IP address pattern then Session
Manager uses the location administered for the SIP Entity. In this sample configuration
Locations are added to SIP Entities (Section 6.5), so it was not necessary to add a pattern.
The following screen shows the addition of SessionManager, this location will be used for
Session Manager. Click Commit to save.
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Note: Call bandwidth management parameters should be set per customer requirement.
Repeat the preceding procedure to create a separate Location for CS1000E and Avaya SBCE.
Displayed below is the screen for CS1K-Location used for CS1000E.
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Below is the screen for Loc19-ASBCE used for Avaya SBCE.
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6.4. Adaptations
To view or change adaptations, select Routing  Adaptations. Click on the checkbox
corresponding to the name of an adaptation and Edit to edit an existing adaptation, or the New
button to add an adaptation. Click the Commit button after changes are completed.
The following screen shows the adaptations that were available in the sample configuration.
The adapter named CS1K-Adaptation will later be assigned to the SIP Entity linking Session
Manager to CS1000E for calls involving Cincinnati Bell eVantage IP Service. This adaptation
uses the CS1000Adapter to convert digits between CS1000E and Cincinnati Bell. The Module
parameter fromto=true will include the FROM and TO headers in the digit conversion.
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Scrolling down, in the Digit Conversion for Incoming Calls to SM section, click Add to
configure entries for calls from CS1000E users to Cincinnati Bell. The text below and the screen
example that follows explain how to use Session Manager to convert the CS1000E directory
numbers that are in the From and P-Asserted-Identity headers to the corresponding Cincinnati
Bell DID numbers.
Matching Pattern
Min
Max
Delete Digits
Insert Digits
Address to modify
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Enter Avaya CS1000E extensions (or extension ranges via
wildcard pattern matching). For other entries, enter the dialed
prefix for any SIP endpoints registered to Session Manager (if
any).
Enter minimum number of digits (e.g., 4).
Enter maximum number of digits (e.g., 4).
Enter 0, unless digits should be removed from dialed number
before routing by Session Manager. For CS1000E extensions
that do not match the last digits of the Cincinnati Bell DID, enter
the number of digits in the extension to remove all digits.
Enter the Cincinnati Bell DID corresponding to the matched
extension or DID prefix for a range of extensions.
Select both.
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Scrolling down, the following screen shows a portion of the CS1K-Adaptation adapter that can
be used to convert digits between the CS1000E extension numbers and the DID numbers
assigned by Cincinnati Bell.
An example portion of the settings for Digit Conversion for Outgoing Calls from SM (i.e.,
inbound to CS1000E) is shown below. It can be observed that the first two entries are used to
match a range of numbers while the last entry is used to match on a specific number.
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6.5. Add SIP Entities
A SIP Entity must be added for Session Manager and for each SIP telephony system connected
to it which includes CS1000E and Avaya SBCE. Navigate to Routing  SIP Entities in the
left-hand navigation pane and click on the New button in the right pane (not shown).
In the General section, enter the following values. Use default values for all remaining fields:
Name:
Enter a descriptive name.
FQDN or IP Address: Enter the FQDN or IP address of the SIP Entity that is used for SIP
signaling.
Type:
Enter Session Manager for Session Manager, CM for
CS1000E and SIP Trunk for Avaya SBCE.
Adaptation:
This field is only present if Type is not set to Session Manager.
If applicable, select the Adaptation Name that will be applied to
this entity.
Location:
Select one of the locations defined previously.
Time Zone:
Select the time zone for the location above.
The following screen shows the addition of Session Manager. The IP address of the Session
Manager signaling interface is entered for FQDN or IP Address.
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To define the ports used by Session Manager, scroll down to the Port section of the SIP Entity
Details screen. This section is only present for Session Manager SIP entities. This section
defines a default set of ports that Session Manager will use to listen for SIP requests, typically
from registered SIP endpoints. Session Manager can also listen on additional ports defined
elsewhere such as the ports specified in the SIP Entity Link definition in Section 6.6.
In the Port section, click Add and enter the following values. Use default values for all
remaining fields:
Port:
Port number on which Session Manager can listen for SIP
requests.
Protocol:
Transport protocol to be used to send SIP requests.
Default Domain:
The domain used for the enterprise.
Defaults can be used for the remaining fields. Click Commit to save.
For the compliance test, four Port entries were added.
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The following screen shows the addition of CS1000E. The FQDN or IP Address field is set to
the IP address of the Node IP on CS1000E defined in Section 5.1.1. The Adaptation field is set
to the CS1K-Adaptation created in Section 6.4 and the Location is set to the one defined for
CS1000E in Section 6.3.
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The following screen shows the addition of Avaya SBCE SIP Entity. The FQDN or IP Address
field is set to the IP address of its private network interface (see Figure 1). The Location is set to
the one defined for Avaya SBCE in Section 6.3. Link Monitoring Disabled was selected for
SIP Link Monitoring.
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6.6. Add Entity Links
A SIP trunk between Session Manager and a telephony system is described as an Entity Link.
Two Entity Links were created; one to CS1000E and one to Avaya SBCE. To add an Entity
Link, navigate to Routing  Entity Links in the left-hand navigation pane and click on the
New button in the right pane (not shown). Fill in the following fields in the new row that is
displayed:
Name:
SIP Entity 1:
Protocol:
Port:
SIP Entity 2:
Port:
Trusted:
Enter a descriptive name.
Select the SIP Entity for Session Manager.
Select the transport protocol used for this link.
Port number on which Session Manager will receive SIP requests from the
far-end.
Select the name of the other system. For CS1000E, select the CS1000E
SIP Entity defined in Section 6.5.
Port number on which the other system receives SIP requests from the
Session Manager.
Check this box. Note: If this box is not checked, calls from the associated
SIP Entity specified in Section 6.5 will be denied.
Click Commit to save. The following screens illustrate the Entity Links to CS1000E and Avaya
SBCE.
Entity Link to CS1000E:
Entity Link to Avaya SBCE:
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6.7. Add Routing Policies
Routing policies describe the conditions under which calls will be routed to the SIP Entities
specified in Section 6.5. Two routing policies must be added; one for CS1000E and one for
Avaya SBCE. To add a routing policy, navigate to Routing  Routing Policies in the left-hand
navigation pane and click on the New button in the right pane (not shown). The screen below is
displayed. Fill in the following:
In the General section, enter the following values. Use default values for all remaining fields:
Name:
Enter a descriptive name.
Notes:
Add a brief description (optional).
In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not
shown). Select the appropriate SIP entity to which this routing policy applies and click Select
(not shown). The selected SIP Entity displays on the Routing Policy Details page as shown
below. Use default values for remaining fields. Click Commit to save.
The following screens show the Routing Policies for CS1000E and Avaya SBCE.
Routing Policy for CS1000E:
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Routing Policy for Avaya SBCE:
6.8. Add Dial Patterns
Dial Patterns are needed to route calls through Session Manager. For the compliance test, dial
patterns were needed to route calls from CS1000E to Cincinnati Bell and vice versa. Dial
Patterns define which route policy will be selected for a particular call based on the dialed digits,
destination domain and originating location. To add a dial pattern, navigate to Routing  Dial
Patterns in the left-hand navigation pane and click on the New button in the right pane (not
shown). Fill in the following, as shown in the screens below:
In the General section, enter the following values. Use default values for all remaining fields:
Pattern:
Enter a dial string that will be matched against the Request-URI of the
call.
Min:
Enter a minimum length used in the match criteria.
Max:
Enter a maximum length used in the match criteria.
SIP Domain:
Enter the destination domain used in the match criteria.
Notes:
Add a brief description (optional).
In the Originating Locations and Routing Policies section, click Add. From the Originating
Locations and Routing Policy List that appears (not shown), select the appropriate originating
location for use in the match criteria. Lastly, select the routing policy from the list that will be
used to route all calls that match the specified criteria. Click Select.
Default values can be used for the remaining fields. Click Commit to save.
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Two examples of the dial patterns used for the compliance test are shown below. The first
example shows that that in the shared test environment, 11 digit dialed numbers that begin with 1
originating from CS1K-Location uses route policy To-ASBCE.
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The second example shows that a 10 digit number starting with 51355551 and originating from
Loc19-ASBCE uses route policy To-CS1K. This is a DID range 513-555-5100 through 513555-5199 assigned to the enterprise from Cincinnati Bell.
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6.9. Add/Verify Avaya Aura® Session Manager Instance
The creation of a Session Manager Instance provides the linkage between System Manager and
Session Manager. This was most likely done as part of the initial Session Manager installation.
To add a Session Manager, navigate to Elements  Session Manager  Session Manager
Administration in the left-hand navigation pane and click on the New button in the right pane
(not shown). If the Session Manager instance already exists, click View (not shown) to view the
configuration. Enter/verify the data as described below and shown in the screen below:
In the General section, enter the following values:
SIP Entity Name:
Select the SIP Entity created for Session
Manager.
Description:
Add a brief description (optional).
Management Access Point Host Name/IP: Enter the IP address of the Session Manager
management interface.
The screen below shows the Session Manager values used for the compliance test.
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In the Security Module section, enter the following values:
SIP Entity IP Address:
Should be filled in automatically based on the SIP Entity
Name. Otherwise, enter IP address of Session Manager
signaling interface.
Network Mask:
Enter the network mask corresponding to the IP address of
Session Manager.
Default Gateway:
Enter the IP address of the default gateway for Session
Manager.
Use default values for the remaining fields. Click Save (not shown) to add this Session Manager.
The screen below shows the remaining Session Manager values used for the compliance test.
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7. Configure Avaya Session Border Controller for Enterprise
This section covers the configuration of the Avaya SBCE. It is assumed that the Avaya SBCE
software has already been installed.
A pictorial view of this configuration is shown below. It shows the components needed for the
compliance test. Each of these components is defined in the Avaya SBCE web configuration as
described in the following sections.
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Use a WEB browser to access the Element Management Server (EMS) web interface, and enter
https://<ip-addr>/ucsec in the address field of the web browser, where <ip-addr> is the
management LAN IP address of the Avaya SBCE.
Log in with the appropriate credentials. Click Sign In.
The main page of the UC-Sec Control Center will appear.
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To view system information that was configured during installation, navigate to UC-Sec Control
Center  System Management. A list of installed devices is shown in the right pane. In the
case of the sample configuration, a single device named ASBCE is shown. To view the
configuration of this device, click the monitor icon as shown below.
The System Information screen shows the Network Settings, DNS Configuration and
Management IP information provided during installation and corresponds to Figure 1. The Box
Type was set to SIP and the Deployment Mode was set to Proxy. Default values were used for
all other fields.
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7.1. Network Management
The Network Management screen is where the network interface settings are configured and
enabled. During the installation process of Avaya SBCE, certain network-specific information is
defined such as device IP address(es), public IP address(es), netmask, gateway, etc. to interface
the device to the network. It is this information that populates the various Network
Management tab displays, which can be edited as needed to optimize device performance and
network efficiency. Navigate to UC-Sec Control Center  Device Specific Settings 
Network Management and verify the IP addresses assigned to the interfaces and that the
interfaces are enabled. The following screen shows the private interface is assigned to A1 and the
external interface is assigned to B1.
The following screen shows interface A1 and B1 are Enabled. To enable an interface click its
Toggle State button.
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7.2. Routing Profile
Routing profiles define a specific set of packet routing criteria that are used in conjunction with
other types of domain policies to identify a particular call flow and thereby ascertain which
security features will be applied to those packets. Parameters defined by Routing Profiles include
packet transport settings, name server addresses and resolution methods, next hop routing
information, and packet transport types.
Create a Routing Profile for Session Manager and Cincinnati Bell eVantage IP Service. To add a
routing profile, navigate to UC-Sec Control Center  Global Profiles  Routing and select
Add Profile. Enter a Profile Name and click Next to continue.
In the new window that appears, enter the following values (not shown). Use default values for
all remaining fields:
URI Group:
Next Hop Server 1:
Next Hop Server 2:
Routing Priority Based on
Next Hop Server:
Outgoing Transport:
Select “*” from the drop down box.
Enter the Domain Name or IP address of the
Primary Next Hop server.
(Optional) Enter the Domain Name or IP address of
the secondary Next Hop server.
Checked.
Choose the protocol used for transporting outgoing
signaling packets.
Click Finish.
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In the shared test environment the following screen shows the Routing Profile to Session
Manager. The Next Hop Server 1 IP address must match the IP address of Session Manager
Entity created in Section 6.5. The Outgoing Transport is set to TCP and matched the Protocol
set in the Session Manager Entity Link for Avaya SBCE in Section 6.6.
The following screen shows the Routing Profile to Cincinnati Bell. In the Next Hop Server 1
field enter the IP address and port that Cincinnati Bell uses to listen for SIP traffic. Enter UDP
for the Outgoing Transport field.
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7.3. Topology Hiding Profile
The Topology Hiding profile manages how various source, destination and routing information
in SIP and SDP message headers are substituted or changed to maintain the integrity of the
network. It hides the topology of the enterprise network from external networks.
Create a Topology Hiding Profile for the enterprise and Cincinnati Bell eVantage IP Service. In
the sample configuration, the Enterprise and CBT SIP Trunk profiles were cloned from the
default profile. To clone a default profile, navigate to UC-Sec Control Center  Global
Profiles  Topology Hiding. Select the default profile and click on Clone Profile as shown
below.
Enter a descriptive name for the new profile and click Finish.
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Edit the Enterprise profile to overwrite the headers shown below to the enterprise domain. The
Overwrite Value should match the Domain set in Session Manager (Section 6.2). Click Finish
to save the changes.
It is not necessary to modify the CBT SIP Trunk profile from the default values. The following
screen shows the Topology Hiding Policy created for Cincinnati Bell.
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7.4. Server Interworking Profile
The Server Internetworking profile configures and manages various SIP call server-specific
parameters such as TCP and UDP port assignments, heartbeat signaling parameters (for HA
deployments), DoS security statistics, and trusted domains. Interworking Profile features are
configured based on different Trunk Servers. There are default profiles available that may be
used as is, or modified, or new profiles can be configured as described below.
In the sample configuration, separate Server Interworking Profiles were created for Enterprise
and Cincinnati Bell.
7.4.1. Server Interworking Profile – Enterprise
To create a new Server Interworking Profile for the enterprise, navigate to UC-Sec Control
Center  Global Profiles  Server Interworking and click on Add Profile as shown below.
Enter a descriptive name for the new profile and click Next to continue.
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In the new window that appears, enter the following values. Use default values for all remaining
fields:
Hold Support: Select RFC2543 - c=0.0.0.0.
T.38 Support: Checked.
Click Next to continue.
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Default values can be used for the next two windows that appear. Click Next to continue.
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On the Advanced Settings window uncheck the following default settings:
Topology Hiding: Change Call-ID
Change Max Forwards
Click Finish to save changes.
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7.4.2. Server Interworking Profile – Cincinnati Bell
To create a new Server Interworking Profile for Cincinnati Bell, navigate to UC-Sec Control
Center  Global Profiles  Server Interworking and click on Add Profile as shown in the
previous section. Enter a descriptive name for the new profile and click Next to continue.
In the new window that appears, enter the following values. Use default values for all remaining
fields:
Hold Support:
T.38 Support:
None
Checked.
Click Next to continue.
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Default values can be used for the next two windows that appear. Click Next to continue.
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On the Advanced Settings window uncheck the following default settings:
Topology Hiding: Change Call-ID
Change Max Forwards
Click Finish to save changes.
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7.5. Server Configuration
The Server Configuration screen contains four tabs: General, Authentication, Heartbeat, and
Advanced. Together, these tabs configure and manage various SIP call server-specific
parameters such as TCP and UDP port assignments, heartbeat signaling parameters, DoS security
statistics, and trusted domains.
In the sample configuration, separate Server Configurations were created for Session Manager
and Cincinnati Bell.
7.5.1. Server Configuration – Session Manager
To add a Server Configuration Profile for Session Manager, navigate to UC-Sec Control Center
 Global Profiles  Server Configuration and click on Add Profile (not shown). Enter a
descriptive name for the new profile and click Next.
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In the new window that appears, enter the following values. Use default values for all remaining
fields:
Server Type:
IP Addresses /
Supported FQDNs:
Supported Transports:
TCP Port:
Select Call Server from the drop-down box.
Enter the IP address of Session Manager. This should
match the IP address of the SIP Entity for Session Manager
in Section 6.5.
Select the transport protocol used to create the Avaya
SBCE Entity Link in Session Manager in Section 6.6.
Port number on which to send SIP requests to Session
Manager. This should match the port number used in the
Avaya SBCE Entity Link in Session Manager in Section
6.6.
Click Next to continue.
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Verify Enable Authentication is unchecked as Session Manager does not require authentication.
Click Next to continue.
In the new window that appears, enter the following values. Use default values for all remaining
fields:
Enabled Heartbeat:
Method:
Frequency:
From URI:
TO URI:
Checked.
Select OPTIONS from the drop-down box.
Choose the desired frequency in seconds the Avaya
SBCE will send SIP OPTIONS to Session Manager. For
compliance testing 120 seconds was chosen.
Enter an URI to be sent in the FROM header for
SIP OPTIONS.
Enter an URI to be sent in the TO header for SIP
OPTIONS.
Click Next to continue.
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In the new window that appears, select the Interworking Profile created for the enterprise in
Section 7.4.1. Use default values for all remaining fields. Click Finish to save the configuration.
7.5.2. Server Configuration - Cincinnati Bell
To add a Server Configuration Profile for Cincinnati Bell, navigate to UC-Sec Control Center
 Global Profiles  Server Configuration and click on Add Profile (not shown). Enter a
descriptive name for the new profile and click Next.
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In the new window that appears, enter the following values. Use default values for all remaining
fields:
Server Type:
IP Addresses /
Supported FQDNs:
Supported Transports:
UDP Port:
Select Trunk Server from the drop-down box.
Enter the IP address(es) of the SIP proxy(ies) of the service
provider. In the case of the compliance test, this is the IP
address of the Cincinnati Bell eVantage IP Service. This
will associate the inbound SIP messages from Cincinnati
Bell to this Sever Configuration.
Select the transport protocol to be used for SIP traffic
between Avaya SBCE and Cincinnati Bell. For compliance
testing UDP was used.
Enter the port number that Cincinnati Bell uses to
send SIP traffic. For compliance testing 5060 was used.
Click Next to continue.
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If using trunk registration, select Enable Authentication. Enter the user name provided by
Cincinnati Bell in the User Name field. Leave the Realm blank to have it detected from the
server challenge. Enter the password provided by Cincinnati Bell in the Password field. Click
Next to continue.
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In the new window that appears, enter the following values. Use default values for all remaining
fields:
Enabled Heartbeat:
Method:
Frequency:
From URI:
TO URI:
Checked.
If using trunk registration, select REGISTER from the
drop-down box. Otherwise, select OPTIONS.
Choose the desired frequency in seconds the Avaya SBCE
will send REGISTER/OPTIONS messages to Cincinnati
Bell. For compliance testing 120 seconds was chosen.
Enter an URI to be sent in the FROM header for
SIP REGISTER/OPTIONS. In the example below
5135555180@192.168.62.92 was used.
Enter an URI to be sent in the TO header for SIP
REGISTER/OPTIONS. In the example below
5135555180@192.168.176.168 was used.
Click Next to continue.
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In the new window that appears, select the Interworking Profile created for Cincinnati Bell in
Section 7.4.2. Use default values for all remaining fields. Click Finish to save the configuration.
7.6. Media Rule
Media Rules define RTP media packet parameters such as prioritizing encryption techniques and
packet encryption techniques. Together these media-related parameters define a strict profile that
is associated with other SIP-specific policies to determine how media packets matching these
criteria will be handled by the Avaya SBCE security product.
Create a custom Media Rule to set the Quality of Service and Media Anomaly Detection. The
sample configuration shows a custom Media Rule New-Low-Med was created for Cincinnati
Bell eVantage IP Service and the enterprise.
To create a custom Media Rule, navigate to UC-Sec Control Center  Domain Policies 
Media Rules. With default-low-med selected, click Clone Rule as shown below.
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Enter a descriptive name for the new rule and click Finish.
On the Media QoS tab select the proper Quality of Service (QoS). Avaya SBCE can be
configured to mark the Differentiated Services Code Point (DSCP) in the IP Header with specific
values to support Quality of Service policies for the media. The following screen shows the QoS
values used for compliance testing.
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7.7. Signaling Rule
Signaling Rules define the action to be taken (Allow, Block, Block with Response, etc.) for each
type of SIP-specific signaling request and response message. When SIP signaling packets are
received by Avaya SBCE, they are parsed and “pattern-matched” against the particular signaling
criteria defined by these rules. Packets matching the criteria defined by the Signaling Rules are
tagged for further policy matching.
Clone and modify the default signaling rule to remove unnecessary SIP headers and add the
proper quality of service to the SIP message. To clone a signaling rule, navigate to UC-Sec
Control Center  Domain Policies  Signaling Rules. With the default rule chosen, click on
Clone Rule (not shown). Enter a descriptive name for the new rule and click Finish.
In the sample configuration, signaling rule Avaya was created for Session Manager to prevent
certain headers in the SIP messages sent from the CS1000E and Session Manager from being
propagated to Cincinnati Bell. Select this rule in the center pane, then select the Request
Headers tab to view the manipulations performed on the request messages such as the initial
INVITE or UPDATE message. The following screen shows the Alert-Info, P-Location, and xnt-e164-clid headers removed during the compliance test.
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Similarly, manipulations can be performed on the SIP response messages. These can be viewed
by selecting the Response Headers tab as shown below.
On the Signaling QoS tab select the proper Quality of Service (QoS). The Avaya SBCE can be
configured to mark the Differentiated Services Code Point (DSCP) in the IP Header with specific
values to support Quality of Service policies for signaling. The following screen shows the QoS
values used for compliance testing.
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A separate signaling rule SIPTrunk Sig Rule was created for Cincinnati Bell eVantage IP
Service by cloning the default signaling rule and changing the Signaling QoS parameters as
shown below.
7.8. Application Rule
Application Rules define which types of SIP-based Unified Communications (UC) applications
the Avaya SBCE security device will protect: voice, video, and/or Instant Messaging (IM). In
addition, you can determine the maximum number of concurrent voice and video sessions the
network will process in order to prevent resource exhaustion.
Create an Application Rule to increase the number of concurrent voice traffic. The sample
configuration cloned and modified the default application rule to increase the number of
Maximum Concurrent Session and Maximum Sessions Per Endpoint. To clone an
application rule, navigate to UC-Sec Control Center  Domain Policies  Application
Rules. With the default rule chosen, click on Clone Rule (not shown). Enter a descriptive name
for the new rule and click Finish.
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Modify the rule by clicking the Edit button. The following screen shows the modified
Application Rule with the Maximum Concurrent Sessions and Maximum Session Per
Endpoint set to 2000. Set the values high enough for the amount of traffic the network is able
process. Keep in mind Avaya SBCE takes 30 seconds for sessions to be cleared after disconnect.
7.9. Endpoint Policy Group
The rules created within the Domain Policy section are assigned to an Endpoint Policy Group.
The Endpoint Policy Group is then applied to a Server Flow in Section 7.12. Create a separate
Endpoint Policy Group for the enterprise and the Cincinnati Bell eVantage IP Service.
To create a new policy group, navigate to UC-Sec Control Center  Domain Policies 
Endpoint Policy Groups and click on Add Group as shown below.
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The following screen shows Enterprise_DomPolicy created for the enterprise. Set the
Application, Media, and Signaling rules to the ones previously created for the enterprise. Set
the Border, Security and Time of Day rules to default or default-low.
The following screen shows SIP Trunk_DomPolicy created for Cincinnati Bell. Set the
Application, Media, and Signaling rules to the one previously created for Cincinnati Bell. Set
the Border, Security, and Time of Day rules to default or default-high.
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7.10. Media Interface
The Media Interface screen is where the SIP media ports are defined. Avaya SBCE will listen for
SIP media on the defined ports. Create a SIP Media Interface for both the inside and outside IP
interfaces.
To create a new Signaling Interface, navigate to UC-Sec Control Center  Device Specific
Settings  Media Interface and click Add Media Interface.
The following screen shows the media interfaces created in the sample configuration for the
inside and outside IP interfaces.
After the media interfaces are created, an application restart is necessary before the changes will
take effect. Navigate to UC-Sec Control Center  System Management and click the forth
icon from the right to restart the applications as highlighted below.
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7.11. Signaling Interface
The Signaling Interface screen is where the SIP signaling ports are defined. Avaya SBCE will
listen for SIP requests on the defined ports. Create a Signaling Interface for both the inside and
outside IP interfaces. To create a new Signaling Interface, navigate to UC-Sec Control Center
 Device Specific Settings  Signaling Interface and click Add Signaling Interface.
The following screen shows the signaling interfaces created in the sample configuration for the
inside and outside IP interfaces.
7.12. End Point Flows - Server Flow
When a packet is received by Avaya SBCE, the content of the packet (IP addresses, URIs, etc.)
is used to determine which flow it matches. Once the flow is determined, the flow points to a
policy which contains several rules concerning processing, privileges, authentication, routing,
etc. Once routing is applied and the destination endpoint is determined, the policies for this
destination endpoint are applied. The context is maintained, so as to be applied to future packets
in the same flow. The following screen illustrates the flow through the Avaya SBCE to secure a
SIP Trunk call.
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Create a Server Flow for Session Manager and Cincinnati Bell eVantage IP Service. To create a
Server Flow, navigate to UC-Sec Control Center  Device Specific Settings  End Point
Flows. Select the Server Flows tab and click Add Flow as shown in below.
In the new window that appears, enter the following values. Use default values for all remaining
fields:
Flow Name:
Server Configuration:
Received Interface:
Signaling Interface:
to
Media Interface:
End Point Policy Group:
Routing Profile:
Topology Hiding Profile:
Enter a descriptive name.
Select a Server Configuration created in Section 7.5 to
assign to the Flow.
Select the Signaling Interface created in Section 7.11 the
Server Configuration is allowed to receive SIP messages
from.
Select the Signaling Interface created in Section 7.11 used
communicate with the Server Configuration.
Select the Media Interface created in Section 7.10 used to
communicate with the Server Configuration.
Select the policy created in Section 7.9 assigned to the
Server Configuration.
Select the profile created in Section 7.2 the Server
Configuration will use to route SIP messages to.
Select the profile created in Section 7.3 to apply toward the
Server Configuration.
Click Finish to save and exit.
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The following screen shows the Sever Flow for Cincinnati Bell eVantage IP Service:
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The following screen shows the Sever Flow for Session Manager:
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8. Cincinnati Bell eVantage IP Service Configuration
To use Cincinnati Bell eVantage IP Service, a customer must request the service from Cincinnati
Bell using their sales processes. This process can be initiated by contacting Cincinnati Bell via
the corporate web site at www.cincinnatibell.com and requesting information via the online sales
links or telephone numbers.
9. Verification
This section provides verification steps that may be performed in the field to verify that the
solution is configured properly.
9.1. Avaya Communication Server 1000E Verification
This section illustrates sample verifications that may be performed using the Avaya CS1000E
Element Manager GUI.
9.1.1. IP Network Maintenance and Reports Commands
From Element Manager, navigate to System  IP Network  Maintenance and Reports as
shown below. In the resultant screen on the right, click the Gen CMD button.
The General Commands page is displayed as shown below.
A variety of commands are available by selecting an appropriate Group and Command from the
drop-down menus, and selecting Run.
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To check the status of the SIP Gateway to Session Manager in the sample configuration, select
Sip from the Group menu and SIPGwShow from the Command menu. Click Run. The example
output below shows that Session Manager (10.64.19.150, port 5060, TCP) has SIPNPM Status
Active.
The following screen shows a means to view registered SIP telephones. The screen shows the
output of the Command sigSetShowAll in Group SipLine. At the time this screen was
captured, the SIP telephone with DN 7108 was involved in an active call with the Cincinnati Bell
eVantage IP Service.
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The following screen shows a means to view IP UNIStim telephones. The screen shows the
output of the Command isetShow in Group Iset. At the time this screen was captured, the
UNIStim telephone with IP address 10.80.150.111 was involved in an active call with the
Cincinnati Bell eVantage IP Service.
9.1.2. System Maintenance Commands
A variety of system maintenance commands are available by navigating to System 
Maintenance using Element Manager. The user can navigate the maintenance commands using
either the Select by Overlay approach or the Select by Functionality approach.
The following screen shows an example where Select by Overlay has been chosen. The various
overlays are listed, and the LD 96 – D-Channel is selected.
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On the preceding screen, if D-Channel Diagnostics is selected on the right, a screen such as the
following is displayed. D-Channel number 15, which is used in the sample configuration, is
established EST and active ACTV.
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9.2. Avaya Aura® Session Manager Verification
The following steps may be used to verify the Session Manager configuration:
1. Verify the call routing administration on Session Manager by logging in to System
Manager and executing the Call Routing Test. Expand Elements  Session Manager 
System Tools  Call Routing Test. Populate the field for the call parameters of interest.
For example, the following screen shows a call routing test for an outbound call to PSTN
via Cincinnati Bell. Under Routing Decisions, observe the call will rout via Avaya
SBCE to Cincinnati Bell. Scroll down to inspect the details of the Routing Decision
Process if desired (not shown).
2. Verify that endpoints at the enterprise site can place calls to the PSTN and that the call
remains active for more than 35 seconds. This time period is included to verify that
proper routing of the SIP messaging has satisfied SIP protocol timers.
3. Verify that endpoints at the enterprise site can receive calls from the PSTN and that the
call can remain active for more than 35 seconds.
4. Verify that the user on the PSTN can end an active call by hanging up.
5. Verify that an endpoint at the enterprise site can end an active call by hanging up
9.3. Avaya Session Border Controller for Enterprise Verification
This section contains verification steps that may be performed using the Avaya Session Border
Controller for Enterprise.
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9.3.1. Incidents
The Incidents Log Viewer display alerts captured by the Avaya SBCE appliance. Select the
Incidents link along the top of the screen.
The following screen shows an example SIP messages that do not match a Server Flow for an
incoming message.
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9.3.2. Diagnostics
The Diagnostics tool allows for PING tests and displays application and protocol use. Select the
Diagnostics link along the top of the screen.
The following screen shows an example PING to Session Manager from the internal signaling
interface of the Avaya SBCE.
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9.3.3. Trace Settings
The Trace Settings tool is for configuring and displaying call traces and packet captures for the
Avaya SBCE. Navigate to Troubleshooting  Trace Settings as shown below. The following
screen shows an example packet capture on interface A1 with a Maximum Number of Packets
to Capture set to 1000. The Capture Filename CBT-A1.pcap will be created once the Start
Capture button is pressed.
The following screen shows a completed packet capture.
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The packet capture file can be downloaded and viewed using a Network Protocol Analyzer like
Wireshark:
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10. Conclusion
These Application Notes describe the configuration necessary to connect Avaya Communication
Server 1000E, Avaya Aura® Session Manager, and Avaya Session Border Controller for
Enterprise to the Cincinnati Bell eVantage IP Service. The Cincinnati Bell eVantage IP Service
is a SIP-based Voice over IP solution for customers ranging from small businesses to large
enterprises. The Cincinnati Bell eVantage IP Service provides businesses a flexible, cost-saving
alternative to traditional hardwired telephony trunks.
11. Additional References
This section references the documentation relevant to these Application Notes. Additional
Avaya product documentation is available at http://support.avaya.com.
[1] Avaya Communication Server 1000E Installation and Commissioning, November 2010,
Document Number NN43041-310.
[2] Feature Listing Reference Avaya Communication Server 1000, November 2010, Document
Number NN43001-111, 05.01.
[3] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/
[4] Signaling Server IP Line Applications Fundamentals Avaya Communication Server 1000,
Document Number NN43001-125, 03.09 October 2011
[5] Installing and Configuring Avaya Aura® System Platform, Release 6.2.0, March 2012.
[6] Administering Avaya Aura® System Platform, Release 6.2.0, February 2012.
[7] Implementing Avaya Aura ® System Manager, Release 6.2, March 2012
[8] Installing Service Packs for Avaya Aura® Session Manager, February 2012, Document
Number 03-603863
[9] Implementing Avaya Aura® Session Manager, February 2012, Document Number 03-603473.
[10] Linux Platform Base and Applications Installation and Commissioning Avaya Communication
Server 1000, Document Number NN43001-315, 05.18 January 2012
[11] SIP Software for Avaya 1100 Series IP Deskphones-Administration , Document Number
NN43170-600, Standard 04.02 December 2011
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©2012
Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at devconnect@avaya.com.
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