Application Notes for Avaya IP Office Release 8.1 and Avaya

Application Notes for Avaya IP Office Release 8.1 and Avaya
Avaya Solution & Interoperability Test Lab
Application Notes for Avaya IP Office Release 8.1 and
Avaya Session Border Controller for Enterprise 6.2, with
AT&T IP Toll Free Service – Issue 1.0
Abstract
These Application Notes describe the steps for configuring Avaya IP Office R8.1 and the Avaya
Session Border Controller for Enterprise 6.2, with the AT&T IP Toll Free service using AVPN
or MIS/PNT transport connections.
The Avaya Session Border Controller for Enterprise is the point of connection between Avaya
Avaya IP Office and the AT&T IP Toll Free service and is used to not only secure the SIP
trunk, but also to make adjustments to the SIP signaling for interoperability.
Avaya IP Office 8.1 is a versatile communications solution that combines the reliability and
ease of a traditional telephony system with the applications and advantages of an IP telephony
solution. This converged communications solution can help businesses reduce costs, increase
productivity, and improve customer service.
The AT&T IP Toll Free service is a managed Voice over IP (VoIP) communications solution
providing toll-free services over SIP trunks for business customers.
AT&T is a member of the Avaya DevConnect Service Provider program. Information in these
Application Notes has been obtained through compliance testing and additional technical
discussions. Testing was conducted via the DevConnect Program.
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Table of Contents
1. Introduction ............................................................................................................................. 4
2. General Test Approach and Test Results ................................................................................ 4
2.1. Interoperability Compliance Testing ................................................................................ 4
2.2. Test Results ...................................................................................................................... 5
2.2.1. Known Limitations ................................................................................................... 5
2.3. Support ............................................................................................................................. 6
3. Reference Configuration ......................................................................................................... 6
3.1. Illustrative Configuration Information ............................................................................. 9
3.2. Call Flows ........................................................................................................................ 9
3.2.1.
Inbound ................................................................................................................... 9
3.2.2.
Coverage to Voicemail ......................................................................................... 10
4. Equipment and Software Validated ...................................................................................... 11
5. Avaya IP Office Configuration ............................................................................................. 11
5.1. Physical, Network, and Security Configuration ............................................................. 12
5.2. Licensing ........................................................................................................................ 13
5.3. System Settings .............................................................................................................. 14
5.3.1. System Tab.............................................................................................................. 14
5.3.2. LAN 1 Settings ....................................................................................................... 15
5.3.3. LAN 2 Settings ....................................................................................................... 18
5.3.4. Voicemail ................................................................................................................ 18
5.3.5. System Telephony Configuration ........................................................................... 19
5.3.6. System Codecs Configuration................................................................................. 20
5.4. SIP Line .......................................................................................................................... 21
5.4.1. SIP Line - Transport Tab ........................................................................................ 22
5.4.2. SIP Line - SIP URI Tab .......................................................................................... 23
5.4.3. SIP Line - VoIP Tab ............................................................................................... 25
5.4.4. SIP Line - T38 Fax .................................................................................................. 26
5.5. Users, Extensions, and Hunt Groups .............................................................................. 27
5.5.1. Analog User Extn207 .............................................................................................. 27
5.5.2. IP Phone User Extn500 ........................................................................................... 28
5.5.3. SIP Telephone Users (Avaya 1120E and Avaya IP Office SoftPhone).................. 31
5.5.4. Hunt Groups ............................................................................................................ 37
5.6. Short Codes .................................................................................................................... 40
5.6.1. Call Center Access to Voicemail Pro...................................................................... 40
5.6.2. Voicemail Access.................................................................................................... 40
5.7. Incoming Call Routes ..................................................................................................... 41
5.8. Call Center Provisioning in Avaya Voicemail Pro ........................................................ 42
5.9. SIP Options Frequency................................................................................................... 44
5.10.
Saving Configuration Changes to Avaya IP Office.................................................... 45
6. Configure Avaya Session Border Controller for Enterprise ................................................. 46
6.1. Initial Installation/Provisioning ...................................................................................... 46
6.2. Log into the Avaya SBCE .............................................................................................. 46
6.3. Global Profiles................................................................................................................ 47
6.3.1. Server Interworking – to Avaya IP Office .............................................................. 47
6.3.2. Server Interworking – to AT&T ............................................................................. 49
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6.3.3. Routing – to Avaya IP Office ................................................................................. 49
6.3.4. Routing – to AT&T ................................................................................................. 51
6.3.5. Server Configuration – To Avaya IP Office ........................................................... 51
6.3.6. Server Configuration – To AT&T .......................................................................... 53
6.3.7. Topology Hiding – Avaya IP Office ....................................................................... 54
6.3.8. Topology Hiding – AT&T ...................................................................................... 55
6.3.9. Signaling Manipulation ........................................................................................... 56
6.4. Domain Policies ............................................................................................................. 56
6.4.1. Application Rules.................................................................................................... 56
6.4.2. Media Rules ............................................................................................................ 57
6.4.3. Signaling Rules ....................................................................................................... 58
6.4.4. Endpoint Policy Groups – Avaya IP Office ............................................................ 60
6.4.5. Endpoint Policy Groups – AT&T ........................................................................... 60
6.5. Device Specific Settings................................................................................................. 61
6.5.1. Network Management ............................................................................................. 61
6.5.2. Advanced Options ................................................................................................... 62
6.5.3. Media Interfaces...................................................................................................... 63
6.5.4. Signaling Interface .................................................................................................. 64
6.5.5. Endpoint Flows – to Avaya IP Office ..................................................................... 64
6.5.6. Endpoint Flows – To AT&T ................................................................................... 65
7. AT&T IP Toll Free Service Configuration ........................................................................... 66
8. Verification Steps.................................................................................................................. 66
8.1. AT&T IP Toll Free Service ............................................................................................ 67
8.2. Avaya IP Office 6.2 ........................................................................................................ 67
8.2.1. System Status Application ...................................................................................... 67
8.2.2. System Monitor Application ................................................................................... 69
8.3. Avaya Session Border Controller for Enterprise 6.2...................................................... 71
8.3.1. System Information ................................................................................................. 71
8.3.2. Avaya SBCE Protocol Traces ................................................................................. 72
8.4. Protocol Trace Examples ............................................................................................... 74
9. Conclusion ............................................................................................................................ 75
10. References .......................................................................................................................... 76
11. Addendum 1 – Redundancy to Multiple AT&T Border Elements .................................... 77
12. Appendix: Avaya IP Office 8.1 SIP Line Template .......................................................... 81
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1. Introduction
These Application Notes describe the steps for configuring Avaya IP Office R8.1 (Avaya IP
Office) and the Avaya Session Border Controller for Enterprise 6.2, (Avaya SBCE), with the
AT&T IP Toll Free service using AVPN or MIS/PNT transport connections.
Avaya IP Office is a versatile communications solution that combines the reliability and ease of
a traditional telephony system with the applications and advantages of an IP telephony solution.
This converged communications solution can help businesses reduce costs, increase productivity,
and improve customer service.
The Avaya Session Border Controller for Enterprise is the point of connection between Avaya
Avaya IP Office and the AT&T IP Toll Free service and is used to not only secure the SIP trunk,
but also to make adjustments to the SIP signaling for interoperability.
The AT&T IP Toll Free service is a managed Voice over IP (VoIP) communications solution
providing toll-free services over SIP trunks for business customers. The AT&T Toll Free service
utilizes AVPN1 or MIS/PNT2 transport services.
Note – The AT&T IP Toll Free service will be referred to as IPTF in the remainder of this
document.
2. General Test Approach and Test Results
The test environment (see Figure 1) consists of:
A simulated enterprise with Avaya IP Office, Avaya IP Office telephones and fax machines
(Ventafax application), and the Avaya SBCE.
Laboratory versions of the IPTF service, to which the simulated enterprise was connected via
AVPN/MIS transport.
The test objectives were to verify the features and functionality described 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
The interoperability compliance testing focused on verifying inbound call flows (see Section
3.2) between Avaya IP Office, the Avaya SBCE, and the IPTF service.
1
2
AVPN uses compressed RTP (cRTP).
MIS/PNT does not support cRTP.
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The compliance testing was based on a test plan provided by AT&T, for the functionality
required for certification as a solution supported on the AT&T network. Calls were made from
PSTN across the IPTF service network.
The following SIP trunking VoIP features were tested with the IPTF service:
Incoming calls from PSTN, routed by the IPTF service, via the Avaya SBCE, to Avaya
IP Office. These calls are via the Avaya IP Office SIP Line and may be
generated/answered by Avaya SIP telephones/Softphones, H.323 telephones, Analog
telephones, Analog fax machines or via Hunt Groups. Coverage to Avaya IP Office
Voicemail Pro, and Voicemail Pro auto-attendant applications, may also be used.
Inbound fax using T38 or G.711, and G3 or SG3 endpoints.
Proper disconnect when the caller abandoned a call before answer, and when the Avaya
IP Office party or the PSTN party terminated an active call.
Proper busy tone heard when an Avaya IP Office user called a busy PSTN user, or a
PSTN user called a busy Avaya IP Office user (i.e., if no redirection was configured for
user busy conditions).
SIP OPTIONS monitoring of the health of the SIP trunk. In the reference configuration
Avaya IP Office sent OPTIONS to the IPTF service Border Element and AT&T
responded with 405 Method Not Allowed (which is the expected response). That response
is sufficient for Avaya IP Office to consider the connection up.
Incoming calls using the G.729 and G.711 ULAW codecs.
Long duration calls.
DTMF transmission (RFC 2833) for successful voice mail navigation, including
navigation of a simple auto-attendant application configured on Avaya IP Office
Voicemail Pro as well as IPTF DTMF generated features.
Telephony features such as call waiting, hold, transfer, and conference.
AT&T IP Toll Free features such as Legacy Transfer Connect and Alternate Destination
Routing were also tested.
2.2. Test Results
Interoperability testing of the sample configuration and features described in Section 2.1 were
completed successfully. The following observations were noted during testing:
2.2.1. Known Limitations
1. The Avaya IP Office fax feature “T.38 Fallback” (to G.711) is not supported in the
reference configuration.
2. G.726 codec support - G.726 codec is not supported by IP Office.
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3. Avaya IP Office only supports a packet size of 20 msecs, and therefore does not
specify a PTIME value in the SIP SDP (in either requests or responses) - Network
responses include MAXPTIME=20, and network requests include MAXPTIME=30.
Although no issues were found during testing, the AT&T IPTF service
recommends a value of 30ms when AVPN transport is used.
4. Avaya IP Office uses fixed RFC2833 Telephone Event type 101 in SIP requests –
Avaya IP Office uses a fixed RFC2833 Telephone Event type of 101, and the AT&T
network responses do comply with a value of 101. However, AT&T network SIP
requests specify Telephone Event type 100, and Avaya IP Office complies with a value
of 100.
No issues were found during testing as a result of this behavior.
5. IP Trunk shuffling is not supported with Avaya IP Office 8.1 using the 500v2
platform.
IP Trunk shuffling is supported with Avaya IP Office 8.1 using the server
platform.
2.3. Support
AT&T customers may obtain support for the AT&T IP Toll Free service by calling (800) 3255555.
Avaya customers may obtain documentation and support for Avaya products by visiting
http://support.avaya.com. In the United States, (866) GO-AVAYA (866-462-8292) provides
access to overall sales and service support menus. Customers may also use specific numbers
(provided on http://support.avaya.com) to directly access specific support and consultation
services based upon their Avaya support agreements.
3. Reference Configuration
Note – Documents used to provision the reference configuration are listed in Section 10.
Specific references to these documents are indicated in the following sections by the notation
reference [x], where x is the document reference number.
The reference configuration used in these Application Notes is shown in Figure 1 below and
consists of the following components:
Avaya IP Office provides the voice communications services for a particular enterprise site.
In the reference configuration, Avaya IP Office runs on an IP 500 V2 platform. This solution
is extensible to Avaya IP Office platforms as well (see Item 5 in Section 2.2.1).
Avaya “desk” telephones are represented with an Avaya 1608 H.323 set, an Avaya 6211
Analog set, an Avaya 1120E SIP set, and PC based Avaya IP Office SIP Softphone (in
Default Mode). Fax endpoints are represented by PCs running Ventafax software connected
by modem to an Avaya IP Office analog port.
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Avaya IP Office Voicemail Pro (running on a Windows 2003 server) provided the voice
messaging capabilities in the reference configuration. This solution is extensible to the Avaya
IP Office embedded voice mail as well.
Avaya Session Border Controller for Enterprise running on a CAD-0208 platform. This
solution is extensible to other Avaya Session Border Controller for Enterprise platforms as
well.
UDP/5060 is the recommended transport protocol/port to use on the IP Office LAN1
connection. However TCP/5060 may be used if necessary.
Inbound calls utilize telephone or fax User/Extensions provisioned on Avaya IP Office.
Signaling is sent between Avaya IP Office and the IPTF service via the Avaya SBCE.
The AT&T IPTF service requires the following SIP trunk network settings to the IPTF
Border Element:
o UDP transport using port 5060
o RTP port ranges 16384-32767
The AT&T IP Toll Free service provided the inbound access numbers (DID and DNIS) used
in the reference configuration.
The Avaya IP Office 500V2 platform and the Avaya SBCE CAD-0208 platform used in the
reference configuration deployed using the following configuration (referred to as an IP Office
“one-wire” configuration).
Avaya IP Office LAN1 interface (labeled “LAN”) connected to the CPE private network.
Avaya SBCE A1 interface connected to the CPE private network.
Avaya SBCE B1 interface connected to the AT&T IP Toll Free service network router.
Note – In the reference configuration, the IP Office LAN2 interface, and the Avaya SBCE E3
interface (CAD-0208 platform), are both connected to a separate “management” subnet.
Note – The IP Office “one-wire” configuration described in this document is the preferred
configuration for the IP Office/Avaya SBCE solution. However an IP Office “two-wire”
configuration is also supported.
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Figure 1: Reference Configuration
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3.1. Illustrative Configuration Information
The specific values listed in Table 1 below and in subsequent sections are used in the reference
configuration described in these Application Notes, and are for illustrative purposes only.
Customers must obtain and use the values based on their own specific configurations.
Note - The AT&T IPTF IP addressing shown in this document is an example. AT&T Customer
Care will provide the actual IP addressing as part of the IPTF provisioning process.
Component
Illustrative Value in these
Application Notes
Avaya IP Office 500 V2 Platform
Private IP Address (LAN1 interface, labeled
192.168.42.1
“LAN” on the chassis, see Section 5.1)
LAN2 interface, (labeled “WAN” on the chassis),
192.168.1.22
for management access.
Avaya SBCE CAD-0208 Platform
Private IP Address (A1 interface).
192.168.42.20
Public IP Address (B1 interface).
192.168.64.130
Management IP address (interface labeled “E3”).
192.168.1.20
AT&T IPTF Service
Border Element IP Address
135.25.29.74
AT&T Access router interface
192.168.64.254
Table 1: Illustrative Values Used in these Application Notes
3.2. Call Flows
To understand how inbound AT&T IPTF service calls are handled by Avaya IP Office, two basic
call flows are described in this section.
3.2.1. Inbound
The first call scenario illustrated in the figure below is an inbound AT&T IPTF service call that
arrives on Avaya IP Office, which in turn routes the call to a hunt group, phone or a fax
endpoint.
1.
2.
3.
4.
A PSTN phone originates a call to an IPTF service number.
The PSTN routes the call to the AT&T IPTF service network.
The AT&T IPTF service routes the call to the Avaya SBCE.
The Avaya SBCE performs SIP Network Address Translation (NAT) and any specified
SIP header modifications, and routes the call to Avaya IP Office.
5. Avaya IP Office applies any necessary digit manipulations based upon the DID and
routes the call to a hunt group, phone or a fax endpoint.
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Figure 2: Inbound AT&T IPTF Call
3.2.2. Coverage to Voicemail
The call scenario illustrated in the figure below is an inbound call that is covered to Voicemail.
In the reference configuration, the Voicemail system used is Avaya IP Office Voicemail Pro,
running on a Windows 2003 server.
1. Same as the first call scenario in Section 3.2.1.
2. The Avaya IP Office phone does not answer the call, and the call covers to the external
application Avaya IP Office Voicemail Pro.
Figure 3: Coverage to Voicemail (Voicemail Pro)
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4. Equipment and Software Validated
The following equipment and software was used for the reference configuration described in these
Application Notes.
Equipment/Software
Release/Version
Avaya IP Office 500 V2
R8.1 (63)
Avaya IP Office Manager
10.1 (63)
Avaya 1608 (H.323) Telephone
Ha1608ua1_3200.bin
Avaya 1120E (SIP) Telephone
04.03.12.00
Avaya IP Office Softphone (SIP)
3.2.3.20 (64770)
Avaya 6211 Analog Telephone
Avaya Session Border Controller for Enterprise
6.2.0 Q30
Fax device
Ventafax 6.3
AT&T IPTF Service via MIS/PNT transport service
VNI 26
connections.
Table 2: Equipment and Software Versions
Note - Testing was performed with IP Office 500 R8.1, but it also applies to IP Office
Server Edition R8.1. Note that IP Office Server Edition requires an Expansion IP Office
500 v2 R8.1 to support analog or digital endpoints or trunks.
5. Avaya IP Office Configuration
Note - This section describes attributes of the reference configuration, but is not meant to be
prescriptive. In the following sections, only the parameters that are highlighted in bold text are
applicable to this document. Other parameter values may or may not match based on local
configurations. Additionally, the screen shots referenced in these sections may not be complete at
times. For more information on installing Avaya IP Office consult reference [1].
Avaya IP Office is configured via the Avaya IP Office Manager program. For more information
on Avaya IP Office Manager, consult reference [2]. From the Avaya IP Office Manager PC,
select Start → Programs → Avaya IP Office → Manager to launch the Manager application.
Enter the appropriate credentials.
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5.1. Physical, Network, and Security Configuration
This section describes attributes of the reference configuration, but is not meant to be
prescriptive.
In the reference configuration the Avaya IP Office 500 V2 platform contained a
COMBO6210/ATM4 module. The COMBO6210/ATM4 is used to add a combination of ports
to an IP500 V2 control unit. The module supports 10 voice compression channels. Codec support
is G.711mu, G729a, G.723 with 64ms echo cancellation and G.722 (supported by Avaya IP
Office Release 8.0 and higher). The module also supports 6 Digital Station ports for digital
stations in slots 1-6 (except 3800, 4100, 4400, 7400, M and T-Series), 2 Analog Extension ports
in slots 7-8, and 4 Analog Trunk ports in slots 9-12.
The following screen shows the Avaya IP Office module configuration used in the reference
configuration. In the screen below, IP 500 V2 is selected in the Group pane, revealing additional
information about the IP 500 V2 in the Details pane.
The Avaya IP Office 500 V2 has two Ethernet ports on the back of the chassis, labeled LAN
(LAN1) and WAN (LAN2). In the reference configuration, the LAN port (LAN1) is connected
to the private CPE network, as described in Section 3. The Avaya SBCE, as well as H.323
telephones, SIP telephones, and the management/Softphone PC used in the reference
configuration, are also connected to the private CPE network. The WAN port (LAN2) is
connected to the CPE management LAN subnet.
Provisioning for these interfaces is described in Section 5.3.2 and 5.3.3.
In order for the Avaya IP Office system to be able to route data to/from the AT&T network, a
default route must be added specifying the Avaya SBCE “A1” interface (e.g., 192.168.42.20). To
add an IP Route in Avaya IP Office, right-click IP Route from the Navigation pane, and select
New. To view or edit an existing route, select IP Route from the Navigation pane, and select the
appropriate route from the Group pane. The following screen shows the relevant default route
using Destination  LAN1.
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5.2. Licensing
The configuration and features described in these Application Notes require the Avaya IP Office
system to be licensed appropriately. If a desired feature is not enabled or there is insufficient
capacity, contact an authorized Avaya sales representative.
To verify that there is a SIP Trunk Channels License with sufficient capacity, click License in
the Navigation pane and SIP Trunk Channels in the Group pane. Confirm a valid license with
sufficient “Instances” (trunk channels) in the Details pane.
If Avaya IP Telephones will be used, verify the Avaya IP endpoints license. Click License in
the Navigation pane and Avaya IP endpoints in the Group pane. Confirm a valid license with
sufficient “Instances” in the Details pane. Note that in some cases duplicate license entries may
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be listed (e.g, Avaya IP endpoints below). One will display a key sequence in the License Key
field while the other will display “Virtual”.
The following screen shows the availability of a valid license for Power User features
(OPTIONAL). In the reference configuration, the user with extension 500 will be configured as
a “Power User”.
5.3. System Settings
This section illustrates the configuration of system settings. Select System in the Navigation
pane to configure these settings.
5.3.1. System Tab
With the proper system name selected in the Group pane, select the System tab in the Details
pane. The following screen shows a portion of the System tab. The Name field can be used for
a descriptive name of the system. In this case, the default system serial number is used as the
name. The Avaya HTTP Clients Only and Enable SoftPhone HTTP Provisioning boxes are
checked to facilitate Avaya IP Office Softphone usage.
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5.3.2. LAN 1 Settings
In the reference configuration, LAN1 was used to connect the Avaya IP Office to the CPE
network (see the note in Section 3). To view or configure the IP address, select the LAN1 tab
followed by the LAN Settings tab, and enter the following:
IP Address: Set to 192.168.42.1 as used in the reference configuration.
DHCP Mode is also set to Server so that IP phones will get an IP Address from the
Avaya IP Office Server. Other parameters on this screen may be set according to
customer requirements.
Click the OK button (not shown).
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Select the LAN1 VoIP tab as shown in the following screen. The following settings were used
in the reference configuration:
The H323 Gatekeeper Enable box is checked to allow the use of Avaya IP Telephones
using the H.323 protocol, such as the Avaya 1600-Series Telephones used in the
reference configuration.
The SIP Trunks Enabled box is checked to support SIP trunking.
The SIP Registrar Enable box is checked to allow Avaya 11xx (SIP) and Avaya IP
Office Softphone (SIP) usage.
RTP Port Number Range: The AT&T IPTF service requires that the RTP use the port
range 16384 to 32767.
o 16384 is entered in the Port Range (Minimum) field.
o 32766 is entered in the Port Range (Maximum) field, as this field requires even
numbers. See Section 6.5.3 for more information on the RTP settings.
DiffServ Settings (optional): If desired, Avaya IP Office can be configured to mark the
Differentiated Services Code Point (DSCP) in the IP Header with specific values to
support Quality of Service policies. The default values were used in the reference
configuration.
Note that on this interface, RTP Keepalives/Scope is set to Disabled (default).
Other parameters on this screen may be set according to customer requirements.
Click the OK button (not shown).
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Select the LAN1  Network Topology tab as shown in the following screen, and enter the
following:
Public IP Address: The 0.0.0.0 default value is used. This means Avaya IP Office will
use the LAN1 IP address specified on the LAN1 LAN Settings tab described above
(192.168.42.1).
Public Port to 5060.
Firewall/NAT Type is set to Open Internet. With this configuration, STUN will not be
used.
Binding Refresh Time is set to 120 (used for OPTIONS interval, see Section 5.9).
Click the OK button (not shown).
Note: The Firewall/NAT Type parameter may need to be different, depending on the type of
firewall or Network Address Translation device used at the customer premise.
If SIP endpoints are used, select the SIP Registrar tab. The following screen shows the settings
used in the reference configuration. Note that the Domain Name field is set to customerb.com,
(the CPE domain), otherwise the LAN1 IP address is used for registration.
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5.3.3. LAN 2 Settings
As described in Section 3, the LAN2 interface was not used for VoIP traffic in the “one-wire”
configuration implemented in the reference configuration. However the LAN2 interface was
used to manage the IP Office platform and was connected to a separate management subnet.
IP Address: In the reference configuration the IP Office management address is
192.168.1.22.
Other parameters on this screen were set to defaults.
On the VoIP tab verify that the H.323 Gatekeeper, SIP Trunks, and SIP Registrar boxes
are not checked.
The Network Topology tab uses default values.
Click the OK button (not shown).
5.3.4. Voicemail
As described in Sections 1 and 2, Avaya IP Office Voicemail Pro was used in the reference
configuration, running on a Windows 2003 Server. The installation and provisioning of Avaya
IP Office Voicemail Pro is beyond the scope of this document. See reference [4] & [5] for more
information on installing and provisioning Avaya IP Office Voicemail Pro.
To view or change Avaya IP Office Voicemail settings, select the Voicemail tab as shown in the
following screen. The settings presented here simply illustrate the reference configuration and
are not intended to be prescriptive.
Set Voicemail Type: Voicemail Lite/Pro.
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Set Voicemail IP Address: to the IP address of the platform running Voicemail Pro.
Other parameters on this screen may be set according to customer requirements.
Click the OK button (not shown).
5.3.5. System Telephony Configuration
To view or change telephony settings, select the Telephony tab and Telephony sub-tab as
shown in the following screen. The settings presented here simply illustrate the reference
configuration and are not intended to be prescriptive.
In the reference configuration, the Inhibit Off-Switch Forward/Transfer box is unchecked so
that call forwarding and call transfer to PSTN destinations via the AT&T IPTF service can be
tested.
The Companding Law parameters are set to U-LAW as is typical in North America. Other
parameters on this screen may be set according to customer requirements.
OPTIONAL: The Default Name Priority parameter can be relevant to SIP Trunking. The
option to Favor Trunk or Favor Directory can be set system-wide using the screen below, or
set uniquely for each line. Favor Trunk was used in the reference configuration. With the
option set to Favor Directory, Avaya IP Office will prefer to display names found in a personal
or system directory over those arriving from the far-end, if there is a directory match to the caller
ID. This capability is also defined in the SIP Line tab in Section 5.4. A user’s personal
directory example is shown in Section 5.5.2.
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5.3.6. System Codecs Configuration
Navigate to the System  Codecs tab. On the left, observe the list of Available Codecs. By
selecting codecs in this column, they will appear in the Default Codec Selection  Unused
column. Codecs may be selected from the Unused list and moved to the Selected column by use
of the >> button, thereby making the selected codecs available in other screens where codec
configuration may be performed (e.g., Sip Lines and Extensions).
The up and down arrow buttons are used to order the selected codecs. By default, all IP (SIP and
H.323) lines (including the SIP Line) and extensions will assume the system default Selected
codec list, unless configured otherwise for the specific line or extension. When completed, click
on OK (not shown).
Note - In the reference configuration the System and Extension (see Section 5.5) codec lists
specify G.711mu and G.729A (in that order), and the SIP Line (see Section 5.4.3) offers G.729A
and G.711mu (in that order). In this manner local Avaya IP Office calls (non-SIP trunk calls)
will attempt G.711mu first, and SIP trunk calls will attempt G.729A first.
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5.4. SIP Line
The SIP Line tab in the Details pane is shown below for Line Number 17, used for the SIP
Trunk to the Avaya SBCE, and ultimately AT&T. Note, if no SIP Line exists, right click on the
Line item in the Navigation pane and select New  SIP Line. SIP Line 17 will be the first SIP
Line number created. The SIP Line form is completed as follows:
ITSP Domain Name: Set to the AT&T border element IP address supplied by AT&T
(e.g., 135.25.29.74).
In Service and Check OOS: These boxes are checked (default).
o Note that the Out Of Service (OOS) option is used in conjunction with SIP
OPTIONS (see Section 5.9).
Call Routing Method: Matched values based on the Request URI, or To Header
contents, may be selected. In the reference configuration, the default Request URI
setting was used.
Country Code: Use the default <blank>.
Send Caller ID: Set to Diversion Header.
REFER Support: Verify that this open is not selected (default).
Use the default values for the other fields.
Click OK.
Optional: As described in Section 5.3.5, the Name Priority parameter may retain the default
System Default setting, or can be specifically configured to Favor Trunk or Favor Directory.
The default System Default setting was used in the reference configuration (see Section 5.3.5
for the System setting of Favor Trunk).
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5.4.1. SIP Line - Transport Tab
Select the SIP Line  Transport tab and configure the following:
ITSP Proxy Address: Set to the Avaya SBCE A1 interface IP address 192.168.42.20
(see Section 3).
Network Configuration  Layer 4 Protocol. Set to UDP.
Network Configuration  Send Port: Set to 5060 (default).
Network Configuration  Use Network Topology Info: Set to LAN 1.
Calls Route via Registrar: Enabled (default).
Click OK (not shown).
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5.4.2. SIP Line - SIP URI Tab
Select the SIP Line  SIP URI tab. On this form a list of the DNIS digits delivered by AT&T
is created. To add a new SIP URI, click the Add… button. In the bottom of the screen, a New
Channel area will be opened. Entries may be specified in two ways:
1. A “wild-card” entry that will use the contents of SIP headers containing “calling”
information.
Note - When this method is used, the inbound AT&T DNIS digits must be specified for
an Avaya IP Office User or Hunt Group on its corresponding SIP tab (see Section 5.5).
Otherwise the call may be denied.
In this method the following information is specified:
The VIA field will automatically be populated with the IP address of the system LAN
interface (LAN 1) with which the SIP trunk is associated (see Sections 5.3.2 and
5.4.1).
Local URI, Contact, Display Name, and PAI: Set to Use Internal Data.
Note – This PAI setting directs Avaya IP Office to send the PAI (P-AssertedIdentity) header instead of the default PPI (P-Preferred-Identity) header when
appropriate (e.g., privacy calls). The PAI header will be populated from the data
set in the SIP tab of the call initiating User as shown in Section 5.5.
Registration: Set to the default 0: <None>.
Incoming Group: Set here to 100. This value references the Incoming Call Routes
in Section 5.7.
Outgoing Group: Set to 100. Note that although the AT&T IP Toll Free service does
not support outbound calls, this value was set to match the Incoming Group.
Max Calls per Channel: In the reference configuration this was set to 10. This sets
the maximum number of simultaneous calls that can use the URI before Avaya IP
Office returns busy to any further calls.
Click OK to save the information.
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2. A specific entry that will match inbound DNIS digits from AT&T.
Note – This method must be used for Avaya IP Office call destinations other than Users
or Hunt Groups, (e.g., Auto Attendant or direct calls to Voicemail Pro), or the calls will
be denied.
In this method the following information is specified:
Local URI, Contact, and Display Name: Set to an AT&T DNIS number (e.g.,
0000041053 to Voicemail Pro).
PAI: Set to Use Internal Data.
Registration: Set to the default 0: <None>.
Incoming Group: Set here to 100. This value references the Incoming Call Routes
in Section 5.7.
Outgoing Group: Set to 100. Note that although the AT&T IP Toll Free service does
not support outbound calls, this value was set to match the Incoming Group.
Max Calls per Channel: In the reference configuration to this was set to 10. This
sets the maximum number of simultaneous calls that can use the URI before Avaya IP
Office returns busy to any further calls.
Click OK to save the information.
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To edit an existing entry, click an entry in the list and click the Edit…button.
When all SIP URI entries have been added/edited, click OK at the bottom of the screen (not
shown).
5.4.3. SIP Line - VoIP Tab
Select the VoIP tab. The Codec Selection drop-down box  System Default will list all
available codecs. Custom was selected, and G729(a) 8K CS-ACELP, and G.711 ULAW 64K
were specified. This will cause Avaya IP Office to include these codecs in the Session
Description Protocol (SDP) offer, and in the order specified. Note that in the reference
configuration G.729A is set as the preferred codec on the connection to the AT&T IPTF
network.
T.38 fax was used in the reference configuration. Set the Fax Transport Support dropdown menu to T38. Note that the T.38 Fallback option is not supported in the reference
configuration (see Section 2.2.1). Note that Error Correction Mode (ECM) is enabled by
default on the T.38 Fax tab (Section 5.4.4). ECM is supported by the AT&T IPTF
service. G.711 fax also worked in the reference configuration (T.38 option disabled);
however T.38 is the preferred method.
The DTMF Support parameter can remain set to the default value RFC2833.
The Re-invite Supported parameter can be checked to allow for codec re-negotiation in
cases where the target of an incoming call or transfer does not support the codec
originally negotiated on the trunk.
Click OK (not shown).
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Note - By default the VoIP Silence Suppression box is not checked. This disables the use of the
G.729B codec. If silence suppression is desired, check this box, and enable the VoIP Silence
Suppression option on the Extension form VoIP tab (see Section 5.5.2).
5.4.4. SIP Line - T38 Fax
Note - The settings on this tab are only accessible if Re-invite Supported and a Fax Transport
Support option are selected on the VoIP tab (Section 5.4.3).
Note - All default values were used in the reference configuration. Therefore the Use Default
Values box is checked. If different settings are needed, uncheck this box to unlock the form.
Note - Since the AT&T IPTF service does not require registration, the SIP Credentials tab need
not be visited.
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5.5. Users, Extensions, and Hunt Groups
In this section, examples of Avaya IP Office Users, Extensions, and Hunt Groups will be
illustrated. To add a User, right click on User in the Navigation pane, and select New. To edit
an existing User, select User in the Navigation pane, and select the appropriate user to be
configured in the Group pane.
5.5.1. Analog User Extn207
The following screen shows the User tab for User Extn207. This user corresponds to the Avaya
Analog 6211 set.
The following screen shows the SIP tab for User Extn207 (use the arrow buttons in the upper
right corner to navigate to the SIP tab). The SIP Name and Contact parameters are configured
with the associated AT&T DNIS number of the user, (e.g., 0000031053). These parameters
configure the user part of the SIP URI in the From header for outgoing SIP headers, and allow
matching of the SIP URI for incoming calls, without having to enter this number as an explicit
SIP URI for the SIP Line (see Section 5.4.2).
The SIP Display Name (Alias) parameter can optionally be configured with a descriptive name.
If all calls involving this user and a SIP Line should be considered private, then the Anonymous
box may be checked to withhold the user’s information from the network.
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The following screen shows the Extension information for this user. To view, select Extension
from the Navigation pane, and the appropriate extension from the Group pane (e.g., 207).
5.5.2. IP Phone User Extn500
To create a new extension, right click on Extension from the Navigation pane and select New
and H.323. Alternatively edit an existing extension by selecting an extension in the Group pane.
Step 1 - The following screen shows a 1608 IP Telephone provisioned in the User tab for User
Extn500. In the reference configuration, this user will be granted “Power User” features.
Password: This password is used by user applications such as SoftConsole, Phone
Manager and TAPI. It is also used for users with Dial In access. Note that this is not the
user's phone log in code (see the information on the Telephony tab  Supervisor
Settings below), or their Voicemail mailbox password (see information on the Voicemail
tab below).
The Profile parameter is set to Power User.
The Enable Softphone box is checked, along with other advanced capabilities.
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Step 2 - Like the Analog Extn207 user, the SIP tab (use the arrow buttons in the upper right
corner to navigate to the SIP tab) for User Extn500 is configured with a SIP Name and Contact
specifying the user’s associated AT&T DNIS number (e.g., 0000011051).
Step 3 - The following screen shows the Voicemail tab for User Extn500. The Voicemail On
box is checked, and a Voicemail password can be configured using the Voicemail Code and
Confirm Voicemail Code parameters.
Voice mail navigation and retrieval were performed locally and from PSTN telephones, to test
DTMF using RFC 2833, and to test assignment of an AT&T DNIS number to the “Voicemail
Collect” feature (e.g., via the *17 Short Code shown in Section 5.6). Note that the second
configuration option described in the SIP Line  SIP URI tab (Section 5.4.2) is required for
this type of inbound call to work.
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Step 4 - Select the Telephony  Supervisor Settings tab as shown below. The Login Code
will be used by the telephone user as the phone login password.
Step 5 - The following screen shows the Extension information for this user, to illustrate the
VoIP tab available for an IP Telephone. To view, select Extension from the Navigation pane,
and the appropriate extension (500) from the Group pane. Select VoIP in the Details pane.
Use the IP Address field default value (0.0.0.0).
Note that the same codec list as shown in Section 5.3.6 is used.
Use defaults for the remaining fields.
Note that by default the VoIP Silence Suppression box is not checked (the same applies to
provisioned SIP phones as shown in Section 5.5.3). This disables the use of the G.729B codec. If
silence suppression is desired, check this box, and enable the Silence Suppression option on the
SIP Line form VoIP tab (see Section 5.4.3).
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OPTIONAL: As described in Section 5.3.5, the option to Favor Trunk or Favor Directory can
be set system-wide, or set uniquely for each user based on the SIP Line setting. The systemwide application of Favor Trunk was used in the reference configuration. The following
example shows how the Favor Directory option can be specified for users.
Note – In the reference configuration, the SIP Line is configured with Name Priority  System
Default (see Section 5.4). This setting needs to be changed to Name Priority  Favor
Directory, to enable this feature.
The following screen shows the Personal Directory tab for User Extn500. With the
configuration shown below, if Extn500 receives an inbound AT&T call from the telephone
number 7325552438, the phone will display the name “PSTN Phone” (along with the number),
even if AT&T provided a different name in the SIP INVITE message sent to Avaya IP Office.
5.5.3. SIP Telephone Users (Avaya 1120E and Avaya IP Office
SoftPhone)
In the reference configuration, an Avaya 1120E SIP telephone and Avaya IP Office SoftPhone
were provisioned as SIP users. To create a new extension, right click on Extension from the
Navigation pane and select New and SIP. Alternatively edit an existing extension by selecting an
extension in the Group pane.
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5.5.3.1 SIP Avaya 1120E
Step 1 - The following screen shows an1120E Telephone provisioned in the User tab for User
Extn600.
Password: This password is used by user applications such as SoftConsole, Phone
Manager and TAPI. It is also used for users with Dial In access. Note that this is not the
user's phone log in code (see the information on the Telephony tab  Supervisor
Settings below), or their Voicemail mailbox password (see the Voicemail tab below).
In the reference configuration, the Profile parameter is set to Basic User (default). User
Extn600 does not have the Mobile feature capabilities in the reference configuration.
Step 2 - Like the H.323 Extn500 user, the SIP tab (use the arrow buttons in the upper right
corner to navigate to the SIP tab) for User Extn600 is configured with a SIP Name and Contact
specifying the user’s associated AT&T DNIS number (e.g., 0000021052). Optionally a user can
be set to use privacy for all calls by selecting the Anonymous option.
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Step 3 - Like the H.323 Extn500 user, Extn600 also utilized the external Voicemail Pro server.
The Voicemail On box is checked, and a Voicemail password can be configured using the
Voicemail Code and Confirm Voicemail Code parameters.
Step 4 - Select the Telephony Call Settings tab as shown below. Check the Call Waiting On
box to allow multiple call appearances and transfer operations.
Step 5 - Select the Telephony Supervisor Settings tab as shown below. The Login Code
will be used by the telephone user as the phone login password.
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Step 6 - A new SIP extension may be added by right-clicking on Extension in the Navigation
pane and selecting New and SIP Extension. Alternatively, an existing SIP extension may be
selected in the group pane. The following screen shows the Extn tab for the extension
corresponding to an Avaya 1120E.
The Extension ID is automatically assigned by Avaya IP Office (e.g., 8000). The Base
Extension field is manually populated with the desired extension (e.g., 600).
Ensure the Force Authorization box is checked.
Step 7 - The following screen shows the VoIP tab for extension 600.
The IP Address default value is used (0.0.0.0).
Check the Reserve Avaya IP endpoint license box.
In the reference configuration the Codec Selection parameter is set to Custom, and the
same codec list used in Section 5.3.6 is used.
Other fields may retain default values.
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5.5.3.2 SIP Avaya IP Office Softphone
Repeat the steps shown in Section 5.5.3.1 with the following settings.
Defining a User
User tab
The Avaya IP Office SoftPhone was provisioned as User Extn700.
The Enable Softphone box is checked, the other enabled advanced capabilities
shown (e.g., Power User), are optional.
SIP tab
o SIP Name and Contact specifying the user’s associated AT&T DNIS number
(e.g., 0000031054).
Voicemail tab
o User Extn700 also utilized the embedded Voicemail. The Voicemail On box
is checked, and a Voicemail password can be configured using the Voicemail
Code and Confirm Voicemail Code parameters.
Telephony Call Settings tab.
o Check the Call Waiting On box to allow multiple call appearances and transfer
operations.
Telephony Supervisor Settings tab
o The Login Code is the softphone login password.
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Defining an Extension
SIP Extension tab
o The Extension ID is automatically assigned by Avaya IP Office (e.g., 8004).
The Base Extension field is manually populated with the desired extension
(e.g., 700).
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SIP Extension  VoIP tab for extension 700.
o The IP Address default value is used (0.0.0.0).
o In the reference configuration the Codec Selection parameter is set to
Custom, and the same codec list used in Section 5.3.6 is used.
o Other fields may retain default values.
5.5.4. Hunt Groups
Users may also receive incoming calls as members of a hunt group. To configure a new hunt
group, right-click HuntGroup from the Navigation pane and select New. To view or edit an
existing hunt group, select HuntGroup from the Navigation pane, and the appropriate hunt
group from the Group pane.
Step 1 - The following screen shows the Hunt Group tab for hunt group Sales. This hunt group
was configured to contain the Analog telephone (Extn207), and the SIP Softphone Extn700. In
the reference configuration, these telephones extensions are rung based on idle time, due to the
Ring Mode setting LongestWaiting. Click the Edit button to select/deselect from the User List
included in the Hunt Group from the list of available users.
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Step 2 - Under the Queuing tab, check the Queuing On box and set the Queue Length field to
any desirable value. Use the default values for all the other fields.
Step 3 - Under the Announcements tab, check the Announcements On box. The wait time can
be set to any desirable value. Make sure that the Synchronize Calls box is checked. These
announcements are played if an agent for a particular skill is unavailable.
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Step 4 - The following screen shows the SIP tab for hunt group Sales. The SIP Name and
Contact are configured with the AT&T DNIS number 0000041054. In Section 5.7, an Incoming
Call Route will map 0000041054 to this hunt group.
Similarly, additional hunt groups Parts and Service, are created by following Steps 1-4.
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5.6. Short Codes
Avaya IP Office provides predefined Short Codes, however new Short Codes may be defined to
match number strings to an action. To add a Short Code, right click on Short Code in the
Navigation pane, and select New. To edit an existing Short Code, click Short Code in the
Navigation pane, and the Short Code to be configured in the Group pane.
5.6.1. Call Center Access to Voicemail Pro
In the reference configuration, Call Center functionality is configured on Voicemail Pro (see
Section 5.8). In order to access this functionality, short codes can be used. The following section
shows the short code set to access this functionality.
5.6.2. Voicemail Access
In this case, the Code *17 is defined for Feature  Voicemail Collect. This Short Code will be
used as one means to allow an AT&T DNIS number to be programmed to route directly to voice
messaging, (via inclusion of this Short Code as the destination of an Incoming Call Route in
Section 5.7).
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5.7. Incoming Call Routes
Each Incoming Call Route will map a specific AT&T DNIS number to a destination user, hunt
group, or function on Avaya IP Office. To add an incoming call route, right click on Incoming
Call Route in the Navigation pane, and select New. To edit an existing incoming call route,
select Incoming Call Route in the Navigation pane, and the appropriate incoming call route to
be configured in the Group pane.
In the screen shown below, the incoming call route for Incoming Number  0000011051 is
illustrated.
The Line Group ID is set to 100, matching the Incoming Group field configured in the SIP
URI tab for the SIP Line to the Avaya SBCE/AT&T in Section 5.4.2.
Select the Destinations tab. From the Destination drop-down, select the extension to receive
the call when AT&T delivers DNIS digits 000011051. In the reference configuration DNIS
digits are associated with Avaya IP Office User Extn500 (the 1608 H.323 telephone).
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Repeat the process to route all AT&T DNIS numbers to their associated Avaya IP Office
destinations. For example:
0000001050 *83 (Voicemail Pro Call Center access Short Code)
0000041053 *17 (Voicemail access Short Code)
0000041054  1001 Sales (Sales Hunt Group)
Note that the Destination drop down menu may not contain all desired destinations. In these
cases the desired destination may be manually typed into the Destination field.
5.8. Call Center Provisioning in Avaya Voicemail Pro
In the reference configuration, Avaya Voicemail Pro (running on a Windows 2003 server), is
used for Voicemail processing as well as for basic Call Center functionality.
Note - While Avaya Voicemail Pro provisioning and programming is beyond the scope of this
document, a sample Call Center basic configuration is shown below.
In the reference configuration a Call Center function was provisioned to prompt callers to select
a numeric option (1, 2, or 3), that would forward the call to an associated Avaya IP Office Hunt
Group (Sales, Service, and Parts). This is accomplished via the following steps:
Step 1 – Hunt Groups Sales, Service, and Parts are created in IP Office (Section 5.5.4).
Step 2 – Short Code *83 is created in IP Office for Call Center access (Section 5.6.1).
Step 3 - Incoming Call Route for DNIS digits 0000001050 is defined for access to the Call
Center prompts (Section 5.7).
Step 4 - Via the Voicemail Pro GUI interface on the Voicemail Pro platform:
Open the Voicemail Pro Client application.
Create a Start Point by right clicking on Modules and selecting Add.
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Enter a name (e.g., CallCenter) and click on Ok. The new Start Point “CallCenter” will
appear under Modules, and a Start Point icon will appear in the work area.
Click on the Start Point icon to activate the options bar at the top of the screen. From the
options bar, select the Basic Actions icon
, select the Menu icon
, and click on the
work area to place the Menu icon.
i. Double click the Start Point icon.
1. General tab  Token Name = Start Point
2. Click Ok
ii. Double click the Menu icon.
1. General tab, Token Name = Menu
2. Entry Prompts tab  Create an Entry Prompt that will tell the caller
what digits to press to reach Sales, Service, and Parts (e.g.,
attendant.wav). To modify an existing recording, double click on the
.wav file and rerecord. If no .wav files exist, double click on the
iconto open the .wav editor.
3. Touch Tone tab:
a. Select 1, 2, and 3 as the possible entry digits.
b. Select 4 for No of Retries.
c. Check the Timeout and Invalid Entry options.
4. Click on Ok.
Click on the Telephony Actions icon
, select the Transfer icon
, and click on the
work area to place the Transfer icon. Select and place two more Transfer Icons (these
will be used for Sales, Service, and Parts).
i. Double click on the first Transfer icon (“Sales”)
1. General tab  Token Name = Sales
2. Specific tab  Destination  Mailbox  Sales  Ok
ii. Double click on the second Transfer icon (“Service”).
1. General tab  Token Name = Service
2. Specific tab  Destination  Mailbox  Service  Ok
iii. Double Click on the third Transfer icon (“Parts”).
1. General tab, Token Name = Parts
2. Specific tab, Destination  Mailbox  Parts  Ok
From the options bar, select the Connector icon
and:
i. Drag a connecting flow line from the Start Point box to the Menu box (see
screen shot below).
ii. Drag connecting flow lines from each of the Menu options to their associated
Transfer boxes (see screen shot below).
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Step 5 - From the top menu select File  Save & Make Live, or select the
icon.
When the associated AT&T IPTF number is called from PSTN (e.g., 0000001050), the caller
will be prompted to enter 1, 2, or 3 to access Sales, Service, or Parts. The associated Avaya IP
Office extension (e.g., 207, 500, or 600) will then ring.
5.9. SIP Options Frequency
In the reference configuration, Avaya IP Office periodically checks the health of the SIP Line by
sending a SIP OPTIONS message. In Section 5.4, the SIP Line to the Avaya SBCE/AT&T is
shown with the Check OOS box checked. The Avaya SBCE will pass the OPTIONS message
on to AT&T. If there is no response, Avaya IP Office can mark the trunk out of service. Once the
problem with the SIP Line is resolved, the SIP OPTIONS maintenance will automatically bring
the link back to the in-service state. In addition, for secure networks, the periodic sending of
OPTIONS by Avaya IP Office may serve to keep network Firewall “pinholes” open preventing
the blockage of inbound traffic to Avaya IP Office.
In the reference configuration, Avaya IP Office sourced SIP OPTIONS every 120 seconds, (the
value configured in the Binding Refresh Time provisioned in Section 5.3.2). This interval may
be adjusted as required.
Note – In the reference configuration Avaya IP Office sent OPTIONS to the AT&T IPTF service
Border Element and AT&T responded with 405 Method Not Allowed (which is the expected
response). That response is sufficient for Avaya IP Office to consider the connection up.
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5.10. Saving Configuration Changes to Avaya IP Office
The provisioning changes made in Avaya IP Office Manager must be applied to the Avaya IP
Office server in order for the changes to take effect. At the top of the Avaya IP Office Manager
page click File  Save Configuration (if that option is grayed out, no changes are pending).
A screen similar to the following will appear, with either Merge or Immediate automatically
selected, based on the nature of the configuration changes. The Merge option will save the
configuration change with no impact to the current system operation. The Immediate option will
save the configuration and cause the Avaya IP Office server to reboot.
Click OK to execute the save.
The active configuration may be saved to a file at any time by selecting File  Save
Configuration As.
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6. Configure Avaya Session Border Controller for Enterprise
As described in Section 3, the Avaya SBCE used in the reference configuration ran on a CAD0208 platform. This solution is extensible to other Avaya Session Border Controller for
Enterprise platforms as well.
Note: Only the Avaya SBCE provisioning required for the reference configuration is described
in these Application Notes.
6.1. Initial Installation/Provisioning
Note: The installation and initial provisioning of the Avaya SBCE is beyond the scope of this
document. Refer to reference [7] and [8] for additional information.
IMPORTANT! – During the Avaya SBCE installation, the Avaya SBCE Management
interface, (labeled “E3” on the CAD-0208 platform), must be provisioned on a different
subnet than either of the Avaya SBCE private and public network interfaces (e.g., A1 and
B1). If this is not the case, contact your Avaya representative to have this resolved.
The Avaya SBCE installation typically defines public and private networks. As described in
Section 3, the reference configuration defines the Avaya SBCE private interface A1 (IP address
192.168.42.20) on the same CPE network as the IP Office LAN1 interface (192.168.42.1). The
connection to AT&T used the Avaya SBCE public interface B1 (IP address 192.168.64.130).
6.2. Log into the Avaya SBCE
The follow provisioning is performed via the Avaya SBCE GUI interface, using the “E3”
management LAN connection on the chassis.
A. Access the web interface by typing “https://x.x.x.x” (where x.x.x.x is the management IP
address of the Avaya SBCE).
B. Enter the login ID and password.
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C. The main menu window will open. Note that the installed software version is displayed.
6.3. Global Profiles
Global Profiles allow for configuration of parameters across the Avaya SBCE appliances.
6.3.1. Server Interworking – to Avaya IP Office
Server Interworking allows you to configure and manage various SIP call server-specific
capabilities such as call hold and T.38. This section defines the connection to Avaya IP Office
via the “DMZ” network.
1.
2.
3.
4.
5.
Select Global Profiles from the menu on the left-hand side.
Select Server Interworking.
Select the Add button (not shown) and the Profile name window will open (not shown).
Enter profile name: (e.g., Avaya_SI), and click Next.
The General screen will open.
a. Check T38 Support
b. All other options can be left at default
c. Select Next
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6. On the Privacy/DTMF window (not shown), select Next to accept default values.
7. On the Timers tab SIP Timers/Transport Timers window (not shown), select Next
to accept default values.
8. On the Advanced tab, accept the default values, and click Finish.
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The following screenshot shows the completed General tab form.
6.3.2. Server Interworking – to AT&T
Repeat the steps shown in Section 6.3.1 to add an Interworking Profile for the connection to
AT&T via the public network.
1. Select Global Profiles from the menu on the left-hand side.
2. Select Server Interworking.
3. Select Add Profile.
4. On the General Tab (not shown):
a. Enter a profile name: (e.g., ATT_SI).
b. Check T38 Support.
c. All other options can be left at default.
d. Select Next.
5. At the Privacy tab (not shown), select Next to accept default values.
6. At the Interworking Profile tab (not shown), select Next to accept default values.
7. On the last screen (Advanced options, not shown), accept the default values, and click
Finish.
6.3.3. Routing – to Avaya IP Office
The Routing Profile allows you to manage parameters related to routing SIP signaling messages.
This provisioning defines the Routing Profile for the connection to Avaya IP Office.
1. Select Global Profiles from the menu on the left-hand side.
2. Select the Routing tab (not shown).
3. Select Add Profile (not shown).
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4. Enter Profile Name: (e.g., Avaya_R).
5. Click Next and enter:
a. Next Hop Server 1: 192.168.42.1 (Avaya IP Office LAN1 IP address)
b. Verify Routing Priority Based on Next Hop Server is selected (default).
c. Outgoing Transport: UDP
Note – UDP is the recommended protocol to use on the connection between the
Avaya SBCE and IP Office. However TCP may be used if necessary.
d. Accept remaining default values
6. Click Finish.
The following screenshot shows the completed Routing Profiles form.
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6.3.4. Routing – to AT&T
Repeat the steps in Section 6.3.3 to add a Routing Profile for the connection to AT&T.
1.
2.
3.
4.
5.
Select Global Profiles from the menu on the left-hand side.
Select the Routing tab.
Select Add Profile.
Enter Profile Name: (e.g., ATT_R).
Click Next, then enter the following:
a. Next Hop Server 1: 135.25.29.74 (AT&T Border Element IP address)
b. Verify Routing Priority Based on Next Hop Server is selected (default).
c. Outgoing Transport: UDP
6. Click Finish.
6.3.5. Server Configuration – To Avaya IP Office
This section defines the Server Configuration for the connection to Avaya IP Office. The Server
Configuration screen contains four tabs: General, Authentication, Heartbeat, and Advanced.
Together, these tabs allow you to configure and manage various SIP call server-specific
parameters such as port assignments, IP Server type, heartbeat signaling parameters and some
advanced options.
1. Select Global Profiles from the menu on the left-hand side.
2. Select Server Configuration.
3. Select Add Profile and the Profile Name window will open (not shown). Enter a Profile
Name (e.g., Avaya_SC) and select Next.
4. The Add Server Configuration Profile - General window will Open (not shown).
a. Select Server Type: Call Server
b. IP Address: 192.168.42.1 (Avaya IP Office LAN1 IP Address)
c. Supported Transports: Check UDP
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d. UDP Port: 5060
Note – UDP is the recommended protocol to use on the connection between the
Avaya SBCE and IP Office. However TCP may be used if necessary.
e. Select Next
5. The Add Server Configuration Profile - Authentication window will open (not
shown).
a. Select Next to accept default values.
6. The Add Server Configuration Profile - Heartbeat window will open (not shown).
a. Select Next to accept remaining default values.
7. The Add Server Configuration Profile - Advanced window will open.
a. Select Avaya_SI (created in Section 6.3.1), for Interworking Profile.
b. In the Signaling Manipulation Script field select None.
c. Select Finish.
The following screen shots show the completed General and Advanced tabs.
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6.3.6. Server Configuration – To AT&T
Repeat the steps in Section 6.3.5 to create a Server Configuration for the connection to AT&T.
1. Select Global Profiles from the menu on the left-hand side.
2. Select Server Configuration.
3. Select Add Profile and the Profile Name window will open (not shown). Enter a Profile
Name (e.g., ATT_SC) and select Next.
4. The Add Server Configuration Profile - General window will open (not shown).
a. Select Server Type: Trunk Server
b. IP Address: 135.25.29.74 (AT&T Border Element IP Address)
c. Supported Transports: Check UDP
d. UDP Port: 5060
e. Select Next.
5. The Add Server Configuration Profile - Authentication window will open (not
shown).
a. Select Next to accept default values.
6. The Add Server Configuration Profile - Heartbeat window will open (not shown).
a. Select Next to accept default values.
7. The Add Server Configuration Profile - Advanced window will open.
d. Select ATT_SI (created in Section 6.3.2), for Interworking Profile.
e. In the Signaling Manipulation Script field select None.
a. Select Finish.
The following screen shots show the completed General and Advanced tabs.
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6.3.7. Topology Hiding – Avaya IP Office
The Topology Hiding screen allows you to manage 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.
1.
2.
3.
4.
5.
Select Global Profiles from the menu on the left-hand side.
Select Topology Hiding.
Click default profile and select Clone Profile.
Enter Profile Name: (e.g., Avaya_TH)
For the Header To,
a. In the Criteria column select IP/Domain
b. In the Replace Action column select: Overwrite
c. In the Overwrite Value column: customerb.com
6. For the Header Request Line,
a. In the Criteria column select IP/Domain
b. In the Replace Action column select: Overwrite
c. In the Overwrite Value column: customerb.com
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7. For the Header From,
a. In the Criteria column select IP/Domain
b. In the Replace Action column select: Overwrite
c. In the Overwrite Value column: customerb.com
8. Let the remaining fields default.
9. Click Finish.
6.3.8. Topology Hiding – AT&T
Repeat the steps in Section 6.3.7 to create a Topology Hiding Profile for the connection to
AT&T.
1.
2.
3.
4.
5.
6.
Select Global Profiles from the menu on the left-hand side.
Select Topology Hiding.
Click default profile and select Clone Profile.
Enter Profile Name: (e.g., ATT_TH).
Let all Replace Action default to Auto.
Click Finish.
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6.3.9. Signaling Manipulation
The Avaya SBCE can manipulate inbound and outbound SIP headers. However no SIP header
manipulations were required in the reference configuration.
6.4. Domain Policies
The Domain Policies feature allows you to configure, apply, and manage various rule sets
(policies) to control unified communications based upon various criteria of communication
sessions originating from or terminating in the enterprise. These criteria can be used to trigger
different policies which will apply on call flows, change the behavior of the call, manipulate SIP
headers, and make sure the call does not violate any of the policies.
6.4.1. Application Rules
1.
2.
3.
4.
Select Domain Policies from the menu on the left-hand side
Select the Application Rules
Select the default Rule
Select Clone Rule button
a. Name: default-trunk
b. Click Finish
5. Highlight the rule just created: default-trunk
a. Click the Edit button
b. In the Voice row:
i. Change the Maximum Concurrent Sessions to 2000
ii. Change the Maximum Sessions per Endpoint to 2000
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6.4.2. Media Rules
This Media Rule will be applied to both directions and therefore, only one rule is needed.
1. Select Domain Policies from the menu on the left-hand side menu (not shown).
2. Select the Media Rules (not shown).
3. The Media Rules window will open (not shown). From the Media Rules menu, select the
default-low-med rule
4. Select Clone Rule button
a. Name: default-low-med-QOS
b. Click Finish
5. Highlight the rule just created from the Media Rules menu: default-low-med-QOS
a. Select the Media QOS tab (not shown).
b. Click the Edit button and the Media QOS window will open.
c. Check the Media QOS Marking - Enabled
d. Select the DSCP box
e. Audio: Select AF11 from the drop-down
f. Video: Select AF11 from the drop-down
6. Click Finish
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The next screen shot shows the completed Media Rules window.
6.4.3. Signaling Rules
Signaling Rules may be used to remove or block various SIP headers. However no SIP header
manipulations were required in the reference configuration.
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6.4.3.1 Avaya – Signaling QOS
1. Select Domain Policies from the menu on the left-hand side menu (not shown).
2. Select the Signaling Rules (not shown).
3. The Signaling Rules window will open (not shown). From the Signaling Rules menu,
select the default rule.
4. Select Clone Rule button
o Enter a name: Avaya_SR
o Click Finish
5. Highlight the Avaya_SR rule created in Step 4 and enter the following:
o Select the Signaling QOS tab (not shown).
o Click the Edit button and the Signaling QOS window will open.
o Verify that Signaling QOS is selected.
o Select DCSP.
o Select Value = AF11.
2. Click Finish
6.4.3.2 AT&T – Signaling QOS
1. Select Domain Policies from the menu on the left-hand side menu (not shown).
2. Select the Signaling Rules (not shown).
3. The Signaling Rules window will open (not shown). From the Signaling Rules menu,
select the default rule.
4. Select Clone Rule button
o Enter a name: ATT_SR
o Click Finish
5. Highlight the ATT_SR rule created in Step 4 and enter the following:
o Select the Signaling QOS tab (not shown).
o Click the Edit button and the Signaling QOS window will open.
o Verify that Signaling QOS is selected.
o Select DCSP.
o Select Value = AF11.
3. Click Finish
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6.4.4. Endpoint Policy Groups – Avaya IP Office
1. Select Domain Policies from the menu on the left-hand side
2. Select End Point Policy Groups
3. Select Add Group
a) Name: defaultLowAvaya
b) Application Rule: default-trunk (created in Section 6.4.1)
c) Border Rule: default
d) Media Rule: default-low-med-QOS (created in Section 6.4.2)
e) Security Rule: default-low
f) Signaling Rule: Avaya_SR (created in Section 6.4.3.1)
g) Time of Day: default
4. Select Finish (not shown)
6.4.5. Endpoint Policy Groups – AT&T
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1. Select Domain Policies from the menu on the left-hand side
2. Select End Point Policy Groups
3. Select Add Group
a. Name: defaultLowATT
b. Application Rule: default-trunk (created in Section 6.4.1)
c. Border Rule: default
d. Media Rule: default-low-med-QOS (created in Section 6.4.2)
e. Security Rule: default-low
f. Signaling Rule: ATT_SR (created in Section 6.4.3.2)
g. Time of Day: default
4. Select Finish (not shown)
6.5. Device Specific Settings
The Device Specific Settings feature for SIP allows you to view system information, and
manage various device-specific network parameters such as port ranges.
6.5.1. Network Management
1. Select Device Specific Settings from the menu on the left-hand side
2. Select Network Management
a) The network interfaces are defined during installation. However if these values
need to be modified, do so via this tab.
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3. In addition, the provisioned interfaces may be enabled/disabled via the Interface
Configuration tab.
6.5.2. Advanced Options
In Section 6.5.3, the media UDP port ranges required by AT&T are set (16384 – 32767). By
default this range is already allocated by the Avaya SBCE (22000 - 31000). The following steps
reallocate the port ranges used by the Avaya SBCE.
1. Select Device Specific Settings  Advanced Options from the menu on the left-hand
side.
2. Select the Port Ranges tab.
3. In the Config Proxy Internal Signaling Port Range row, change the range to 42000 –
51000.
4. Scroll to the bottom of the window and select Save (not shown).
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6.5.3. Media Interfaces
The AT&T IPFR-EF service specifies that customers use RTP ports in the range of 16384 –
32767. Both inside and outside ports have been changed but only the outside is required by the
AT&T IPFR-EF service.
1. Select Device Specific Settings from the menu on the left-hand side
2. Select Media Interface
3. Select Add Media Interface
a) Name: Avaya_IPO
b) Media IP: 192.168.42.20 (Avaya SBCE A1 address to IP Office)
c) Port Range: 16384 - 32767
4. Click Finish (not shown)
5. Select Add Media Interface
a) Name: ATT
b) Media IP: 192.168.64.130 (Avaya SBCE B1 address toward AT&T)
c) Port Range: 16384 - 32767
6. Click Finish (not shown)
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6.5.4. Signaling Interface
1. Select Device Specific Settings from the menu on the left-hand side
2. Select Signaling Interface
3. Select Add Signaling Interface
a) Name: Avaya-IPO
b) Media IP: 192.168.42.20 (Avaya SBCE A1 address to IP Office)
c) UDP Port: 5060
Note – UDP is the recommended protocol to use on the connection between the Avaya
SBCE and IP Office. However TCP may be used if necessary.
4. Click Finish
5. Select Add Media Interface
a) Name: ATT
b) Media IP: 192.168.64.130 (Avaya SBCE B1/public address toward AT&T)
c) UDP Port: 5060
6. Click Finish (not shown).
6.5.5. Endpoint Flows – to Avaya IP Office
1.
2.
3.
4.
Select Device Specific Settings from the menu on the left-hand side
Select Endpoint Flows
Select the Server Flows tab
Select Add Flow, and enter the following:
a) Name: Avaya-IPO
b) Server Configuration: Avaya_SC
c) URI Group: *
d) Transport: *
e) Remote Subnet: *
f) Received Interface: ATT
g) Signaling Interface: Avaya-IPO
h) Media Interface: Avaya_IPO
i) End Point Policy Group: defaultLowAvaya
j) Routing Profile: ATT_R
k) Topology Hiding Profile: Avaya_TH
l) File Transfer Profile: None
5. Click Finish (not shown)
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6.5.6. Endpoint Flows – To AT&T
1.
2.
3.
4.
Select Device Specific Settings from the menu on the left-hand side
Select Endpoint Flows
Select the Server Flows tab
Select Add Flow, and enter the following:
a) Name: ATT
b) Server Configuration: ATT_SC
c) URI Group: *
d) Transport: *
e) Remote Subnet: *
f) Received Interface: Avaya-IPO
g) Signaling Interface: ATT
h) Media Interface: ATT
i) End Point Policy Group: defaultLowATT
j) Routing Profile: Avaya_R
k) Topology Hiding Profile: ATT_TH
l) File Transfer Profile: None
5. Click Finish (not shown)
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7. AT&T IP Toll Free Service Configuration
AT&T provides the IPTF service border element IP address, the access DID numbers, and the
associated DNIS digits used in the reference configuration. In addition the AT&T IPTF features,
and their associated access numbers, are also assigned by AT&T. AT&T required that the Avaya
SBCE public (B1) IP address be provided to the IPTF service, as part of the provisioning
process.
8. Verification Steps
The following procedures may be used to verify the Avaya IP Office R8.1 and the Avaya Session
Border Controller for Enterprise 6.2, with the AT&T IP Toll Free service configuration.
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8.1. AT&T IP Toll Free Service
The following scenarios may be executed to verify Avaya IP Office and the Avaya Session
Border Controller for Enterprise 6.2, functionality with the AT&T IPTF service:
Place inbound calls, answer the calls, and verify that two-way talk path exists. Verify that
the calls remain stable for several minutes and disconnects properly.
Verify basic call functions such as hold, transfer, and conference.
Place an inbound call to a telephone, but do not answer the call. Verify that the call
covers to Avaya Aura® Messaging voicemail. Retrieve the message from Avaya Aura®
Messaging either locally or from PSTN.
Using the appropriate IPTF access numbers and codes, verify the “Legacy Transfer
Connect” DTMF initiated features.
Inbound fax using T38 or G.711.
SIP OPTIONS monitoring of the health of the SIP trunk. In the reference configuration
Avaya IP Office sent OPTIONS to the AT&T IPTF service Border Element, (via the
Avaya SBCE), and AT&T responded with 405 Method Not Allowed (which is the
expected response). That response is sufficient for Avaya IP Office to consider the
connection up.
Incoming calls using the G.729 (A or B) and G.711 ULAW codecs.
8.2. Avaya IP Office 8.1
The following items may be used to analyze/troubleshoot Avaya IP Office operations.
8.2.1. System Status Application
The System Status application can be used to monitor or troubleshoot Avaya IP Office. The
System Status application can typically be accessed from Start  Programs  Avaya IP
Office  System Status. The following screen shows an example Logon screen. Enter the
Avaya IP Office IP address in the Control Unit IP Address field, and enter an appropriate User
Name and Password. Click Logon.
After logging in, select Trunks  Line: 17 from the left navigation menu. (SIP Line 17 is
configured in Section 5.4). A screen such as the one shown below is displayed. In the lower left,
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the Trace All button may be pressed to display tracing information as calls are made using this
SIP Line.
The following screen shows an example inbound call where PSTN called Avaya IP Office Hunt
Group “Service” (H.323 phone, Extn500).
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8.2.2. System Monitor Application
The System Monitor application can also be used to monitor or troubleshoot Avaya IP Office
functionality (see reference [3]. The System Monitor application can typically be accessed from
Start  Programs  Avaya IP Office  Monitor.
The Monitor will be active at startup. To pause the Monitor press the Pause
The pause button will be replaced with the Start
button.
button. Press this button to resume the
monitoring. To clear the Monitor display, press the Clear
button.
Below are samples of a monitored inbound call to Avaya IP Office SIP telephone Extn600. The
Monitor will display SIP protocol (first image) as well as internal Avaya IP Office processing
(second image).
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The Monitor application allows the monitored information to be customized. To customize,
select the Options button
that is third from the right in the screen above, or select Filters 
Trace Options. The following screen shows the SIP tab, allowing configuration of SIP
monitoring. In this example, the SIP Rx and SIP Tx boxes are checked. All SIP messages will
appear in the trace with the color blue. To customize the color, right-click on SIP Rx or SIP Tx
and select the desired color.
8.3. Avaya Session Border Controller for Enterprise 6.2
The following items may be used to analyze/troubleshoot Avaya SBCE operations.
8.3.1. System Information
Step 1 - Navigate to System Management  Devices tab.
The software version is shown in the Version column.
Also verify that the Status column says Commissioned.
Step 2 – Click on View (shown above) to display the system information.
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8.3.2. Avaya SBCE Protocol Traces
The Avaya SBCE can take internal traces of specified interfaces. In the example below all SIP
signaling crossing interfaces A1 and B1 are captured.
Step 1 - Navigate to Device Specific Settings  Troubleshooting  Trace
Step 2 - Select the Packet Capture tab and select the following:
Interface : Select A1, B1, or Any from the drop down menu. Note that specifying Any
will capture packets from both the A1 and B1 interfaces used in the reference
configuration.
Local Address: Select All and 5060 from the dropdown menus. This will capture any
packets using port 5060 (e.g., SIP signaling).
Remote Address: Enter *
Protocol: Select All from the dropdown menu.
Maximum Number of Packets to Capture: Specify an amount that will include all the
messaging required for your trace (e.g., 5000). Note that specifying amounts greater than
10,000 may impact system performance.
Capture Filename: Specify a name for the trace file. Include a .pcap extension if you
wish to open the file with Wireshark.
Click Start Capture to begin the trace.
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The capture process will initialize and then display the following status window:
Step 3 – Run the test.
Step 4 - Select Stop Capture button shown above.
Step 5 - Click on the Captures tab and the packet capture is listed as a .pcap file with the date
and time added to filename specified in Step 2 (e.g., test1). Note that the system will
append date/time information to the filename.
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Step 6 - Click on the File Name to download the file and use an application such as Wireshark
to open the trace.
8.4. Protocol Trace Examples
This section shows examples of protocol traces taken at the Avaya SBCE B1 interface (to
AT&T).
The following is an example of an inbound call from AT&T, filtering on the SIP protocol.
The following is an example of a call filtering on DTMF events.
The following is an example of a call filtering on RTP.
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A trace of an idle line can verify that your SIP Trunk from the Avaya SBCE B1 interface
(192.168.64.130) to the IPTF Service border element (135.25.29.74) is up and communicating
with SIP OPTIONS messages and response messages. A SIP 405 Method Not Allowed response
is normal for the Avaya SBCE to AT&T test environment.
9. Conclusion
As illustrated in these Application Notes, Avaya IP Office R8.1 and the Avaya Session Border
Controller for Enterprise 6.2, can be configured to interoperate successfully with the AT&T IP
Toll Free service using AVPN or MIS/PNT transport connections, within the limitations
described in Section 2.2.1.
This solution provides users of Avaya IP Office R8.1 the ability to support inbound calls
utilizing an AT&T IPTF SIP trunk service connection, via AVPN or MIS/PNT transport, using
the platform and service features listed in Section 2.1.
The reference configuration shown in these Application Notes is representative of a basic
enterprise customer configuration and is intended to provide configuration guidance to
supplement other Avaya product documentation. It is based upon formal interoperability
compliance testing as part of the Avaya DevConnect Service Provider program.
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10. References
Avaya:
Avaya product documentation is available at http://support.avaya.com
[1] Avaya IP Office 8.1 Installation,15-601042 Issue 26i – (23 August 2012)
[2] Avaya IP Office R8.1 Manager, 10.115-601011 Issue 29o – (03 August 2012)
[3] Avaya IP Office System Monitor, Document Number 15-601019
[4] Avaya IP Office Voicemail Pro15-601063 Issue 20b - (11 July 2008)
[5] Avaya IP Office Voicemail Pro Example Exercises, Issue 4c (5th May 2004)
[6] Additional Avaya IP Office documentation can be found at:
http://marketingtools.avaya.com/knowledgebase/
[7] Installing Avaya Session Border Controller For Enterprise, Release 6.2, Issue 1, January
2013.
[8] Administering Avaya Session Border Controller, Release 6.2, Issue 1, January 2012
AT&T IPTF Service:
[9] AT&T IP Toll Free Service description http://www.business.att.com/enterprise/Service/business-voip-enterprise/network-based-voipenterprise/ip-toll-free-enterprise//
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11. Addendum 1 – Redundancy to Multiple AT&T Border Elements
AT&T may provide multiple network border elements for redundancy purposes. The Avaya
SBCE can be provisioned to support this redundant configuration.
Given two AT&T border elements 135.25.29.74 and 135.25.29.75, the Avaya SBCE is
provisioned as follows to include the backup trunk connection to 135.25.29.75 (the primary
AT&T trunk connection to 135.25.29.74 is defined in Section 6.3.6).
Step 1: Secondary Location in Server Configuration
1. Select Global Profiles from the menu on the left-hand side
2. Select the Server Configuration
3. Select Add Profile
a) Name: ATT_Secondary
4. On the Add Server Configuration Profile – General tab:
a) Select Server Type: Trunk Server
b) IP Address: 135.25.29.75 (Example Address for a secondary location)
c) Supported Transports: Check UDP
d) UDP Port: 5060
e) Select Next (not shown)
5. On the Authentication tab
a) Select Next to accept defaults (not shown).
6. On the Heartbeat tab:
a) Check Enable Heartbeat
b) Method: OPTIONS
c) Frequency: 60 seconds
d) From URI: secondary@customerb.com
e) To URI: secondary@customerb.com
f) Select Next (not shown)
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7. On the Advanced Tab
a) Click Finish to accept defaults (not shown).
8. Select the Profile created in Section 6.3.6 (e.g., ATT_SC)
9. Select the Heartbeat Tab
10. Select Edit
11. Repeat Steps 6 – 7, but with information for the Primary Trunk as shown below.
Step 2: Add Secondary IP Address to Routing
1.
2.
3.
4.
Select Global Profiles from the menu on the left-hand side
Select the Routing
Select the profile created in Section 6.3.4 (e.g., ATT_R )
Click Edit (not shown)
a) Enter the IP Address of the secondary location in the Next Hop Server 2 (e.g.,
135.25.29.75)
5. Click Finish
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Step 3: Configure End Point Flows – ATT_Secondary
1.
2.
3.
4.
Select Device Specific Settings from the menu on the left-hand side
Select Endpoint Flows
Select the Server Flows Tab
Select Add Flow
a) Name: ATT_Secondary
b) Server Configuration: ATT_Secondary
c) URI Group: *
d) Transport: *
e) Remote Subnet: *
f) Received Interface: Avaya-IPO
g) Signaling Interface: ATT
h) Media Interface: ATT
i) End Point Policy Group: defaultLowATT
j) Routing Profile: Avaya_R
k) Topology Hiding Profile: ATT_TH
l) File Transfer Profile: None
5. Click Finish
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When completed the Avaya SBCE will issue OPTIONS messages to the primary (135.25.29.74)
and secondary (135.25.29.75) border elements.
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12. Appendix: Avaya IP Office 8.1 SIP Line Template
Avaya IP Office Release 8.1 supports a SIP Line Template (in xml format) that can be created
from an existing configuration and imported into a new installation to simplify configuration
procedures as well as to reduce potential configuration errors.
Not all of the configuration information is included in the SIP Line Template, therefore, it is
critical that the SIP Line configuration be verified/updated after a template has been imported,
and additional configuration be supplemented using the settings provided in this Application
Notes.
To create a SIP Line Template from the configuration described in these Application Notes,
configure the parameters as described below.
Create a new registry entry called TemplateProvisioning and set the Value data to 1, as
follows:
Select Start, and then Run. Type regedit as shown below
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Under HKEY_CURRENT_USER, Software, Avaya, IP400, right click on Manager, then
select New, DWORD value, then rename the newly created entry to: TemplateProvisioning.
Right click on the newly created entry and select Modify, change the value under Value Data
from “0” to “1”.
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Reboot the computer.
When the computer comes back up, enable the template by opening IP Office Manager, select
File, and then Preferences. On the Visual Preferences tab, check the Enable Template
Options box, and click OK.
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To create a SIP Line Template from the configuration, on the left Navigation Pane, right click on
the Sip Line (17), and select Generate SIP Trunk Template (not shown)
Enter a descriptive name; ATT was used in the sample template. Note that for ITSP Domain
Name Not Used was used (AT&T uses IP addresses instead of Domain names), an entry is
required here or the template will not run. This entry (Not Used) should be removed after
importing the configuration into a new Avaya IP Office installation.
To generate the template click on Export.
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On the next screen, Template Type Selection, select the Country, enter the name for the
Service Provider, and click Generate Template.
By default the template file is generated to the path \\Program Files\Avaya\IP
Office\Manager\Templates.
The following is an example of the exported SIP Line Template file,
US_AT&T_SIPTrunk.xml:
<?xml version="1.0" encoding="utf-8" ?>
- <Template xmlns="urn:SIPTrunk-schema">
<TemplateType>SIPTrunk</TemplateType>
<Version>20121130</Version>
<SystemLocale>enu</SystemLocale>
<DescriptiveName>ATT</DescriptiveName>
<ITSPDomainName>Not Used</ITSPDomainName>
<SendCallerID>CallerIDDIV</SendCallerID>
<ReferSupport>true</ReferSupport>
<ReferSupportIncoming>2</ReferSupportIncoming>
<ReferSupportOutgoing>2</ReferSupportOutgoing>
<RegistrationRequired>false</RegistrationRequired>
<UseTelURI>false</UseTelURI>
<CheckOOS>true</CheckOOS>
<CallRoutingMethod>1</CallRoutingMethod>
<OriginatorNumber />
<AssociationMethod>SourceIP</AssociationMethod>
<LineNamePriority>FavourTrunk</LineNamePriority>
<UpdateSupport>UpdateNever</UpdateSupport>
<UserAgentServerHeader />
<CallerIDfromFromheader>false</CallerIDfromFromheader>
<PerformUserLevelPrivacy>false</PerformUserLevelPrivacy>
<ITSPProxy>12.40.234.99</ITSPProxy>
<LayerFourProtocol>SipUDP</LayerFourProtocol>
<SendPort>5060</SendPort>
<ListenPort>5060</ListenPort>
<DNSServerOne>0.0.0.0</DNSServerOne>
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<DNSServerTwo>0.0.0.0</DNSServerTwo>
<CallsRouteViaRegistrar>true</CallsRouteViaRegistrar>
<SeparateRegistrar />
<CompressionMode>AUTOSELECT</CompressionMode>
<UseAdvVoiceCodecPrefs>true</UseAdvVoiceCodecPrefs>
<AdvCodecPref>G.711 ULAW 64K,G.729(a) 8K CS-ACELP</AdvCodecPref>
<CallInitiationTimeout>4</CallInitiationTimeout>
<DTMFSupport>DTMF_SUPPORT_RFC2833</DTMFSupport>
<VoipSilenceSupression>false</VoipSilenceSupression>
<ReinviteSupported>true</ReinviteSupported>
<FaxTransportSupport>FOIP_T38</FaxTransportSupport>
<UseOffererPrefferedCodec>false</UseOffererPrefferedCodec>
<CodecLockdown>false</CodecLockdown>
<Rel100Supported>true</Rel100Supported>
<T38FaxVersion>3</T38FaxVersion>
<Transport>UDPTL</Transport>
<LowSpeed>0</LowSpeed>
<HighSpeed>0</HighSpeed>
<TCFMethod>Trans_TCF</TCFMethod>
<MaxBitRate>FaxRate_14400</MaxBitRate>
<EflagStartTimer>2600</EflagStartTimer>
<EflagStopTimer>2300</EflagStopTimer>
<UseDefaultValues>false</UseDefaultValues>
<ScanLineFixup>true</ScanLineFixup>
<TFOPEnhancement>true</TFOPEnhancement>
<DisableT30ECM>false</DisableT30ECM>
<DisableEflagsForFirstDIS>false</DisableEflagsForFirstDIS>
<DisableT30MRCompression>false</DisableT30MRCompression>
<NSFOverride>false</NSFOverride>
</Template>
Next, import the template into the new Avaya IP Office system by creating a new SIP Line as
shown in the screenshot below. In the Navigation Pane on the left, right-click on Line then
navigate to New, New SIP Trunk from Template:
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On the next screen, Template Type Selection, verify that the information in the Country and
Service Provider fields is correct. If more than one template is present, use the drop-down
menus to select the required template. Click Create new SIP Trunk to finish the process.
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©2013
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
DeveloperConnection Program at devconnect@avaya.com.
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