Avaya Solution & Interoperability Test Lab

Application Notes for Configuring Avaya IP Office Release

10.1 to support Telmex SIP Trunking Service using
UDP/RTP - Issue 1.0

Abstract

These Application Notes describe the procedures for configuring Session Initiation Protocol
(SIP) Trunking on an enterprise solution consisting of Avaya IP Office 10.1 to support Telmex
SIP Trunking Service using UDP/RTP.

The test was performed to verify SIP trunk features including basic calls, call forward (all
calls, busy, no answer), call transfer (blind and consult), conference, and voice mail. The calls
were placed to and from the public switched telephone network (PSTN) with various Avaya
endpoints.

Readers should pay attention to Section 2, in particular the scope of testing as outlined in
Section 2.1 as well as the observations noted in Section 2.2, to ensure that their own use cases
are adequately covered by this scope and results.

Telmex is a member of the Avaya DevConnect Service Provider Program. Information in these
Application Notes has been obtained through DevConnect compliance testing and additional
technical discussions. Testing was conducted via the DevConnect Program at the Avaya
Solution and Interoperability Test Lab.

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1. Introduction
These Application Notes describe the steps necessary for configuring Session Initiation Protocol
(SIP) Trunking service between Telmex and an Avaya SIP-enabled enterprise solution.

In the configuration used during the testing, the Avaya SIP-enabled enterprise solution consists
of Avaya IP Office Server Edition Primary Server with an IP500 V2 Expansion System
(hereafter referred to as IP Office), Voicemail Pro and various Avaya endpoints, listed in Section
4.

The Telmex SIP Trunking Service referenced within these Application Notes is designed for
business customers. Customers using this service with the Avaya IP Office solution are able to
place and receive PSTN calls via a broadband wide area network (WAN) connection using the
SIP protocol. This converged network solution is an alternative to traditional PSTN trunks such
as analog and/or ISDN-PRI trunks. This approach generally results in lower cost for the
enterprise.

The terms “service provider” or “Telmex” will be used interchangeably throughout these
Application Notes.

2. General Test Approach and Test Results

The general test approach was to simulate an enterprise site at the Telmex facility in Mexico by
connecting the Avaya SIP-enabled enterprise solution to Telmex’s network, as depicted in Figure
1, and exercise the features and functionalities listed in Section 2.1.

DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent
to the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute full product performance or feature testing performed by
DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a DevConnect member’s solution.

2.1. Interoperability Compliance Testing

To verify SIP trunk interoperability the following features and functionalities were exercised
during the interoperability compliance test:
 Static IP SIP Trunk authentication.
 SIP OPTIONS queries and responses.
 Incoming PSTN calls to various phone types including H.323, SIP, Digital and Analog
telephones at the enterprise. All inbound calls from the PSTN were routed to the
enterprise across the SIP trunk from the service provider.
 Outgoing PSTN calls from various phone types including H.323, SIP, Digital, and
Analog telephones at the enterprise. All outbound PSTN calls were routed from the
enterprise across the SIP trunk to the Service Provider.
 Caller ID presentation.

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  Proper disconnect when the caller abandons the call before the call is answered.
 Proper disconnect via normal call termination by the caller or the called parties.
 Proper disconnect by the network for calls that are not answered (with voicemail off).
 Proper response to busy endpoints.
 Proper response/error treatment when dialing invalid PSTN numbers.
 Calls using the G.729, currently the only codec supported by Telmex.
 No matching codecs.
 Proper early media transmissions.
 Voicemail and DTMF tone support using RFC 2833 (leaving and retrieving voice mail
messages from PSTN phones).
 Outbound Toll-Free calls, interacting with IVR (Interactive Voice Response systems).
 Inbound and outbound PSTN calls to/from Avaya Communicator Softphone client.
 Various call types including: local, long distance, international, etc.
 Caller ID presentation and Caller ID restriction.
 Simultaneous active calls.
 Long duration calls (over one hour).
 User features such as hold and resume, transfer, and conference.
 Off-net call forwarding and mobile twinning.
 Proper response/error treatment to different failure conditions.

Items not supported or not tested included the following:

 Inbound toll-free calls, 911 calls (emergency), “0” calls (Operator), 0+10 digits calls
(Operator Assisted), were not tested.
 T.38 fax is not currently supported by Telmex; therefore T.38 fax was not tested.
 G.711 pass-through fax was not tested.
 SIP REFER is not currently supported by Telmex; therefore SIP REFER was not tested.

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2.2. Test Results
Interoperability testing of Telmex SIP Services was completed with successful results for all test
cases with the exception of the observations/limitations described below.
 Outbound Calling Party Number (CPN) Block: To support outbound privacy calls
(calling party number blocking), Communication Manager sends “anonymous” as the
calling number in the SIP From header, uses the P-Asserted-Identity (PAI) header to pass
the actual calling party number and includes “Privacy: id” in the INVITE. During testing
this type of call resulted in call failure. For security reasons this feature may not be
supported in Mexico.

2.3. Support
For support on Telmex systems visit the corporate Web page at:
http://telmex.com/en/web/empresas

Avaya customers may obtain documentation and support for Avaya products by visiting
http://support.avaya.com. Alternatively, in the United States, (866) GO-AVAYA (866-462-8292)
provides access to overall sales and service support menus.

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3. Reference Configuration
Figure 1 illustrates the test configuration used for the DevConnect compliance testing. The test
configuration simulates an enterprise site with an Avaya SIP-enabled enterprise solution
connected to the Telmex SIP trunking service through a private network connection.

The Avaya components used to create the simulated enterprise customer site includes:
 Avaya IP Office Server Edition Primary Server.
 Avaya IP Office 500 V2 Expansion System.
 Avaya Voicemail Pro for IP Office.
 Avaya 96x1 Series H.323 IP Deskphones.
 Avaya J129 SIP IP Deskphones.
 Avaya Communicator for Windows Softphone.
 Avaya 1408 Digital Deskphone.
 Analog Deskphone.

Located at the simulated enterprise site is the Avaya IP Office Server Edition Primary Server and
the Avaya IP Office 500 V2 expansion system with analog and digital extensions, as well as a
VCM64 (Voice Compression Module) for supporting VoIP codecs. The LAN1 port of the Avaya
IP Office Server Edition Primary Server and the LAN1 port of the Avaya IP Office 500 V2
expansion system were connected to the enterprise LAN (private network). The LAN2 ports
were not used. Endpoints include Avaya 96x1 Series IP Deskphones (with H.323 firmware),
Avaya J129 Deskphones (with SIP firmware), digital, and analog Deskphones, and a PC running
Avaya Communicator for Windows Softphones. The Windows PC was also running Avaya IP
Office Manager to configure and administer the Avaya IP Office system. Mobile Twinning is
configured for some of the Avaya IP Office users so that calls to these user’s extensions will also
ring and can be answered at the configured mobile phones.

In an actual customer configuration, the enterprise site may include additional network
components between the service provider and the Avaya IP Office system, such as firewalls. A
complete discussion of the configuration of these devices is beyond the scope of these
Application Notes. However, it should be noted that all SIP and RTP traffic between the service
provider and the Avaya IP Office system must be allowed to pass through these devices.

Avaya recommends our customers implement Avaya solutions using appropriate security and
encryption capabilities enabled by our products. The testing referenced in this DevConnect
Application Note included the enablement of supported encryption capabilities in the Avaya
products only (private network side). Readers should consult the appropriate Avaya product
documentation for further information regarding security and encryption capabilities supported
by those Avaya products.

Support for these security and encryption capabilities in any non-Avaya solution component is
the responsibility of each individual vendor. Readers should consult the appropriate vendor-
supplied product documentation for more information regarding those products.

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For the testing associated with this Application Note, the interface between the Avaya system
and the Telmex network did not include the use of any specific encryption features, UDP
Transport for signaling and RTP for media was used between the Avaya system and the Telmex
network across the SIP Trunk. TLS transport for signaling and SRTP for media was used inside
of the enterprise (private network side, in between Avaya components inside of the enterprise).

Note – The configuration tasks required to support security and encryption capabilities inside of
the enterprise (private network side) are beyond the scope of these Application Notes; hence they
are not discussed in this document.

Figure 1: Avaya Interoperability Test Lab Configuration

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4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:

Equipment/Software Release/Version
Avaya
Avaya IP Office Server Edition Primary Server 10.1.0.0.0 Build 237
Avaya IP Office 500 V2 expansion system 10.1.0.0.0 Build 237
Avaya IP Office Manager 10.1.0.0.0 Build 237
Avaya Voicemail Pro 10.1.0.0.0 Build 241
Avaya 96x1 Series IP Deskphones (H.323) Version 6.6401
Avaya J129 IP Deskphones (SIP) 1.1.0.0.15
Avaya Communicator for Windows 2.1.4.0 built 256
Avaya Digital Deskphones 1408 R46
Lucent Analog Phone --
Telmex
Cisco CUBE C892fsp 15.3(3)m7
Oracle SBC Unknown
Huawei IMS Unknown

Note: Compliance Testing is applicable when the tested solution is deployed with a standalone
IP Office 500 V2 and also when deployed with all configurations of IP Office Server Edition.

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5. Avaya IP Office Primary Server Configuration
This section describes the Avaya IP Office configuration to support connectivity to the Telmex
SIP Trunk Service. Avaya IP Office is configured through the Avaya IP Office Manager PC
application. From a PC running the Avaya IP Office Manager application, select Start 
Programs  IP Office  Manager to launch the application. Navigate to File  Open
Configuration, select the appropriate Avaya IP Office system from the pop-up window and log
in with the appropriate credentials.

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A management window will appear similar to the one shown on the screen below. If the left
navigation pane does not immediately appear click on Configuration on the top-right of the
screen. All the Avaya IP Office configurable components are shown in the left pane known as
the Navigation Pane. The pane on the right is the Details Pane. These panes will be referenced
throughout the Avaya IP Office configuration. In the reference configuration, IP users are
registered to the Primary server. Digital and Analog users are configured on the IP 500 V2
expansion system. The SIP trunk to Telmex is configured on the Primary server. All licensing
and feature configuration that is not directly related to the interface with the Service Provider
(such as twinning) is assumed to already be in place.

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5.1. Licensing
The configuration and features described in these Application Notes require the 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; first navigate to
iposerver in the Navigation Pane where iposerver is the name of the IP Office Primary Server.
Navigate to System  License in the Navigation Pane. In the Details Pane, verify that the
License Status is Valid and that the number of Instances is sufficient to support the number of
SIP trunk channels.

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5.2. System Settings – Primary Server
Configure the necessary system settings. In an Avaya IP Office the LAN2 tab settings
correspond to the Avaya IP Office WAN port (public network side) and the LAN1 tab settings
correspond to the LAN port (private network side). For the compliance test only the LAN1 port
was used, since the connection to the Telmex SIP trunking service was done through a “private”
network connection within Telmex’s network, and not through a public network connection
(public Internet).

5.2.1. System – LAN1 Settings

To access the LAN1 settings of the IP Office Primary Server, first navigate to System 
iposerver in the Navigation Pane where iposerver is the name of the IP Office Primary Server.
Navigate to the LAN1  LAN Settings tab in the Details Pane. The LAN1 settings for the
compliance testing were configured with following parameters:
 Set the IP Address field to the LAN IP address, e.g., 10.2.20.18.
 Set the IP Mask field to the subnet mask of the enterprise private network, e.g.,
255.255.255.128.
 All other parameters should be set according to customer requirements.
 Click OK to commit (not shown).

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The VoIP tab as shown in the screenshot below was configured with following settings:
 Check the H323 Gatekeeper Enable to allow Avaya IP Telephones/Softphone using the
H.323 protocol to register.
 Select Preferred under H.323 Signaling over TLS. When enabled, TLS is used to
secure the registration and call signaling communication between IP Office and endpoints
that support TLS. The H.323 phones that support TLS are 9608, 9611, 9621, and 9641
running firmware version 6.6 or higher.
 Check the SIP Trunks Enable to enable the configuration of SIP Trunk connecting to
Telmex.
 Check the SIP Registrar Enable to allow Avaya IP Telephones/Softphone to register
using the SIP protocol.
 The SIP Domain Name and SIP Registrar FQDN may be set according to customer
requirements. If the Domain Name is left at the default blank setting, SIP registrations
may use the IP Office LAN1 IP Address.
 Check TLS and verify the TLS Port numbers under Layer 4 Protocol is set to 5061.
 Verify the RTP Port Number Range settings for a specific range for the RTP traffic.
The Port Range (Minimum) and Port Range (Maximum) values were kept as default.
 In the Keepalives section at the bottom of the page, set the Scope field to RTP-RTCP,
Periodic Timeout to 30, and Initial keepalives to Enabled. This will cause the IP Office
to send RTP and RTCP keepalive packets at the beginning of the calls and every 30
seconds thereafter if no other RTP/RTCP traffic is present.
 All other parameters should be set according to customer requirements.
 Click OK to commit (not shown).

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Note: In the compliance test, the LAN1 interface of the IP Office Primary Server was used to
connect IP Office to the enterprise private network (LAN). The LAN1 interface was also used to
connect IP Office to the Telmex SIP trunking service. The LAN2 interface of the IP Office
Primary Server was not used.

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5.2.2. System - Telephony Tab
To access the System Telephony settings, first navigate to System  iposerver in the
Navigation Pane where iposerver is the name of the IP Office Primary Server, navigate to the
Telephony  Telephony tab in the Details Pane, and configure the following parameters:
 Choose the Companding Law typical for the enterprise location, A-Law was used.
 Uncheck the Inhibit Off-Switch Forward/Transfer box to allow call forwarding and
call transfers to the PSTN via the SIP trunk to the service provider.
 All other parameters should be set according to customer requirements.
 Click OK to commit (not shown).

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5.2.3. System - VoIP Tab
For Codec settings, navigate to the iposerver in the Navigation Pane where iposerver is the
name of the IP Office Primary Server, select the VoIP tab and configure the following
parameters:
 Select or enter 101 for RFC2833 Default Payload. This setting was recommended by
Telmex for use with out-of-band DTMF tone transmissions.
 For codec selection, select the codecs and codec order of preference on the right, under
the Selected column. The Default Codec Selection area enables the codec preference
order to be configured on a system-wide basis. The buttons between the two lists can be
used to move codecs between the Unused and Selected lists, and to change the order of
the codecs in the Selected codecs list. By default, all IP phones (SIP and H.323) will use
the system default codec selection shown here, unless configured otherwise for a specific
extension. The example below shows the codecs used for IP phones (SIP and H.323).
 All other parameters should be set according to customer requirements.
 Click OK to commit (not shown).

Note: The codec selections defined under this section (System – VoIP Tab) are the codecs
selected for the IP phones/extensions. The codec selections defined under Section 5.4.5 (SIP
Line – VoIP tab) are the codecs selected for the SIP Line (Trunk).

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5.2.4. System – VoIP Security tab
For the compliance test, SRTP was used internal to the enterprise wherever possible. To view or
configure the media encryption settings, first navigate to System  iposerver in the Navigation
Pane where iposerver is the name of the IP Office Primary Server, select the VoIP Security tab
and configure the following parameters:
 Set the Media drop-down menu to Preferred to have IP Office attempt use encrypted
SRTP for devices that support it and fall back to RTP for devices that do not support
encryption.
 Under Media Security Options, select RTP for the Encryptions and Authentication
fields.
 Confirm that SRTP Window Size is set to 64.
 Under Crypto Suites, select SRTP_AES_CM_128_SHA1_80 and
SRTP_AES_CM_128_SHA1_32
 Click OK to commit (not shown).

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5.3. IP Route
Create an IP route to specify the IP address of the gateway or router where the IP Office needs to
send the packets in order to route calls to Telmex’s network.

 Set the IP Address and IP Mask of the IP subnet of the private side of the Telmex SIP
proxy, or enter 0.0.0.0 to make this the default route.
 Set Gateway IP Address to the IP Address of the default router in the IP Office IP
subnet.
 Set Destination to LAN1 from the pull-down menu.
 Click OK to commit (not shown).

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5.4. SIP Line
A SIP line is needed to establish the SIP connection between Avaya IP Office and the Telmex
service. The recommended method for configuring a SIP Line is to use the template associated
with these Application Notes. The template is an .xml file that can be used by IP Office Manager
to create a SIP Line. Follow the steps in Sections 5.4.1 to create the SIP Line from the template.

Some items relevant to a specific customer environment are not included in the template or may
need to be updated after the SIP Line is created. Examples include the following:
 IP addresses
 SIP Credentials (if applicable)
 SIP URI entries
 Setting of the Use Network Topology Info field on the Transport tab

Therefore, it is important that the SIP Line configuration be reviewed and updated if necessary
after the SIP Line is created via the template. The resulting SIP Line data can be verified against
the manual configuration shown in Section 5.4.2 to 5.4.6.

Alternatively, a SIP Line can be created manually. To do so, first navigate to System 
iposerver in the Navigation Pane where iposerver is the name of the IP Office Primary Server,
right-click on Line in the Navigation Pane and select New  SIP Line. Then, follow the steps
outlined in Sections 5.4.2 to 5.4.6.

5.4.1. Creating a SIP Trunk from an XML Template

DevConnect generated SIP Line templates are always exported in an XML format. These XML
templates do not include sensitive customer specific information and are therefore suitable for
distribution. The XML format templates can be used to create SIP trunks on both IP Office
Standard Edition (500 V2) and IP Office Server Edition systems. Alternatively, binary templates
may be generated. However, binary templates include all the configuration parameters of the
Trunk, including sensitive customer specific information. Therefore, binary templates should
only be used for cloning trunks within a specific customer’s environment.

Copy a previously created template file to a location (e.g., \temp) on the same computer where IP
Office Manager is installed.

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To create the SIP Trunk from the template, navigate to the iposerver in the Navigation Pane
where iposerver is the name of the IP Office Primary Server, right-click on Line in the
Navigation Pane, then navigate to New  New from Template.

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Navigate to the directory on the local machine where the template was copied and select the
template.

After the import is complete, a final import status pop-up window will open stating success or
failure. Click OK

The newly created SIP Line will appear in the Navigation pane (e.g., SIP Line 10).

It is important that the SIP Line configuration be reviewed and updated if necessary after the SIP
Line is created via the template. The resulting SIP Line data can be verified against the manual
configuration shown in Sections 5.4.2 to 5.4.6.

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5.4.2. SIP Line – SIP Line Tab
Navigate to the iposerver in the Navigation Pane where iposerver is the name of the IP Office
Primary Server, on the SIP Line tab in the Details Pane, configure or verify the parameters as
shown below.
 Set the ITSP Domain Name to the Telmex proxy IP address. Note that if this field is left
blank IP Office will insert the ITSP Proxy Address from the transport tab as the ITSP
Domain in the SIP message.
 Set the Local Domain Name to the IP address of the IP Office Primary Server LAN1
port.
 Verify that URI Type is set to SIP.
 Verify that In Service box is checked, which is the default value. This makes the trunk
available to incoming and outgoing calls.
 Verify that Check OOS box is checked, the default value. IP Office will use the SIP
OPTIONS method to periodically check the SIP Line.
 Verify that Refresh Method is set to Auto.
 Verify that Timer (seconds) is set to On Demand.
 Under Redirect and Transfer, set Incoming Supervised REFER and Outgoing
Supervised REFER to Never.
 All other parameters should be set to default or according to customer requirements.
 Click OK to commit (not shown).

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5.4.3. SIP Line - Transport Tab
Select the Transport tab and set the following:
 Set the ITSP Proxy Address to the Telmex proxy IP address.
 Set the Layer 4 Protocol to TLS.
 Set Use Network Topology Info to None (see note below).
 Set the Send Port and Listen Port to 5061.
 Default values may be used for all other parameters.
 Click OK to commit (not shown).

Note – For the compliance testing, the Use Network Topology Info field was set to None, since
no NAT was used in the test configuration. In addition, it was not necessary to configure the
System  LAN1  Network Topology tab for the purposes of SIP trunking. If a NAT is used
between Avaya IP Office and the other end of the trunk, then the Use Network Topology Info
field should be set to the LAN interface (LAN1) used by the trunk and the System  LAN1 
Network Topology tab needs to be configured with the details of the NAT device.

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5.4.4. SIP Line - SIP URI Tab
A SIP URI entry needs to be created to match each incoming number that IP Office will accept
on this line. Select the SIP URI tab, and then click the Add… button and the New Channel area
will appear at the bottom of the pane. To edit an existing entry, click an entry in the list at the
top, and click the Edit… button. In the example screen below, a previously configured entry was
edited. For the compliance test, a single SIP URI entry was created that matched any DID
number assigned to an IP Office user. The entry was created with the parameters shown below:
 Set Local URI, Contact and Display Name to Auto.
 Leave the Header at default P Asserted ID.
 Select P Asserted ID as the Send Caller ID.
 Select None in the Diversion Header drop down menu.
 For Registration, select 0: <None> from the pull-down menu.
 Associate this line with an incoming line group by entering a line group number in the
Incoming Group field. For the compliance test, a new incoming group 10 was defined
that was associated to a single line (line 10).
 Associate this line with an outgoing line group by entering a line group number in the
Outgoing Group field. For the compliance test, a new outgoing group 10 was defined
that was also associated to line 10.
 Set Max Calls per Channel to the number of simultaneous SIP calls that are allowed
using this SIP URI pattern.

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5.4.5. SIP Line - VoIP Tab
Select the VoIP tab, to set the Voice over Internet Protocol parameters of the SIP line. Set or
verify the parameters as shown below:
 The Codec Selection was configured using the Custom option, allowing an explicit
order of codecs to be specified for the SIP Line. The buttons allow setting the specific
order of preference for the codecs to be used on the SIP Line, as shown. Telmex supports
G.729(a) for audio.
 Set Fax Transport Support to None.
 Set the DTMF Support field to RFC2833/RFC4733. This directs Avaya IP Office to
send DTMF tones using RTP events messages as defined in RFC2833.
 Verify Media Security is disabled.
 Check the Re-invite Supported box.
 Check the PRACK/100rel Supported box.
 Default values may be used for all other parameters.
 Click the OK to commit (not shown).

Note: The codec selections defined under this section (SIP Line – VoIP tab) are the codecs
selected for the SIP Line (Trunk). The codec selections defined under Section 5.2.3 (System –
VoIP tab) are the codecs selected for the IP phones/extension (H.323 and SIP).

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5.4.6. SIP Line – SIP Advanced Tab
Select the SIP Advanced tab. For outbound calls with privacy enabled, Avaya IP Office will
replace the calling party number in the From and Contact headers of the SIP INVITE message
with “anonymous”. Avaya IP Office can be configured to use the P-Preferred-Identity (PPI) or P-
Asserted-Identity (PAI) header to pass the actual calling party information for authentication and
billing purposes. By default, Avaya IP Office will use the PPI header for privacy. For the
compliance test, PAI was used for the purposes of privacy.

To configure Avaya IP Office to use the PAI header for privacy calls, on the SIP Advanced tab,
 Check Use PAI for Privacy. All other fields retained their default values.

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5.5. User
Configure the SIP parameters for each user that will be placing and receiving calls via the SIP
Line defined in Section 5.4. To configure these settings, first navigate to the iposerver in the
Navigation Pane where iposerver is the name of the IP Office Primary Server, navigate to User
 Name in the Navigation Pane where Name is the name of the user to be modified. In the
example below, the name of the user is 52204 h3231. Select the SIP tab in the Details Pane. The
values entered for the SIP Name allow matching of the SIP URI for incoming calls without
having to enter this number as an explicit SIP URI for the SIP Line (Section 5.4.4). The SIP
Name and Contact are set to one of the DID numbers assigned to the enterprise by Telmex. 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. This can also be
accomplished by activating Withhold Number on H.323 Deskphones (not shown). Click the OK
to commit (not shown).

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5.6. Outbound Call Routing
For outbound call routing, a combination of system short codes and Automatic Route Selection
(ARS) entries are used. With ARS, features like time-based routing criteria and alternate routing
can be specified so that a call can re-route automatically if the primary route or outgoing line
group is not available. While detailed coverage of ARS is beyond the scope of these Application
Notes, and alternate routing was not used in the reference configuration, this section includes
some basic screen illustrations of the ARS settings used during the compliance testing.

5.6.1. Short Codes and Automatic Route Selection

To create a short code to be used for ARS, first navigate to the iposerver in the Navigation Pane
where iposerver is the name of the IP Office Primary Server, right-click on Short Code on the
Navigation Pane and select New. The screen below shows the short code ? created (default
match), the call is directed to Line Group 50: Main, which is configurable via ARS.
 In the Code field, enter the dial string which will trigger this short code. In this case ?
was used, this character can be used on its own to create a short code match in the absent
of any other short code match.
 Set Feature to Dial. This is the action that the short code will perform.
 Set Telephone Number to “.” The value “.” represents the number dialed by the user
which will be replaced by the full set of dialed digits that triggered the short code match.
This value is passed to ARS.
 Set the Line Group ID to 50: Main to be directed to Line Group 50: Main, which is
configurable via ARS.
 Set the Locale as required, in this case Mexico (Latin Spanish).
 Click the OK to commit (not shown).

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The following screen shows a sample ARS configuration for the route 50: Main.

To create a short code to be used for ARS, first navigate to the iposerver in the Navigation Pane
where iposerver is the name of the IP Office Primary Server, select ARS  50: Main on the
Navigation Pane and click Add (not shown):
 In the Code field, enter the dial string which will trigger this short code. In this case, N;
which represents any digits.
 Set Feature to Dial. This is the action that the short code will perform.
 Set Telephone Number to N”@telmex.com.mx”.
 Set the Line Group ID to the Line Group number being used for the SIP Line, in this
case Line Group ID 10 was used.
 Set the Locale to the specific country and language, in this case Mexico (Latin Spanish).
 Click OK to commit.

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5.7. Incoming Call Route
An incoming call route maps an inbound DID number on a specific line to an internal extension.
To create an incoming call route, first navigate to the iposerver in the Navigation Pane where
iposerver is the name of the IP Office Primary Server, right-click Incoming Call Routes in the
Navigation Pane and select New.

On the Standard tab of the Details Pane, enter the parameters as shown below:
 Set the Bearer Capability to Any Voice.
 Set the Line Group Id to the incoming line group of the SIP line defined in Section 5.4.
 Set the Incoming Number to the incoming number that this route should match.
 Default values can be used for all other fields.

On the Destinations tab, select the destination extension from the pull-down menu of the
Destination field. On completion, click the OK button (not shown). In this example, incoming
calls to the test DID number 1011 on line 2 are routed to extension 52206 anal02.

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5.8. Save Configuration
Navigate to File  Save Configuration in the menu bar at the top of the screen to save the
configuration performed in the preceding sections.

The following will appear, with either Merge or Immediate selected for the Change Mode,
based on the nature of the configuration changes made since the last save. Note that clicking OK
may cause a service disruption. Click OK if desired.

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6. Telmex SIP Trunking Services Configuration
To use the Telmex SIP Trunk service, a customer must request the service from Telmex using
the established sales processes. The process can be started by contacting Telmex via the
corporate web site at: http://telmex.com/en/web/empresas and requesting information.

During the signup process, Telmex and the customer will discuss details about the preferred
method to be used to connect the customer’s enterprise network to the Telmex network. Telmex
will provide IP address and port number, Direct Inward Dialed (DID) numbers to be assigned to
the enterprise, etc. This information is used to complete the Avaya IP Office configuration
discussed in the previous sections.

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7. Verification Steps
This section provides verification steps that may be performed to verify that the solution is
configured properly.

The following steps may be used to verify the configuration:

 Verify that endpoints at the enterprise site can place calls to the PSTN.
 Verify that endpoints at the enterprise site can receive calls from the PSTN.
 Verify that users at the PSTN can end active calls to endpoints at the enterprise by
hanging up.
 Verify that endpoints at the enterprise can end active calls to PSTN users by hanging up.

7.1. IP Office System Status

The following steps can also be used to verify the configuration.

Use the Avaya IP Office System Status application to verify the state of SIP connections.
Launch the application from Start  Programs  IP Office  System Status on the PC
where IP Office Manager is installed, log in with the proper credentials.

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Select the SIP line under Trunks from the left pane. On the Status tab in the right pane, verify
the Current State is Idle for each channel.

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7.2. IP Office Monitor
The Avaya IP Office Monitor application can be used to monitor and troubleshoot signaling
messaging on the SIP trunk. Launch the application from Start  Programs  IP Office 
Monitor on the PC where Avaya IP Office Manager was installed. Click the Select Unit icon on
the taskbar and Select the IP address of the IP Office system under verification.

Clicking the Trace Options icon on the taskbar, selecting the SIP tab allows modifying the
threshold used for capturing events, types of packets to be captured, filters, etc. Additionally, the
color used to represent the packets in the trace can be customized by right clicking on the type of
packet and selecting the desired color.

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8. Conclusion
These Application Notes describe the configuration steps necessary for configuring Session
Initiation Protocol (SIP) Trunk Service for an enterprise solution consisting of Avaya IP Office
Release 10.1 to interoperate with the Telmex SIP Trunking Service, as shown in Figure 1.

The Telmex SIP Trunking Service passed compliance testing with the observations/limitations
outlined in the scope of testing in Section 2.1 as well as under test results in Section 2.2.

9. Additional References
This section references the documentation relevant to these Application Notes. Product
documentation for Avaya IP Office, including the following, is available at:
http://support.avaya.com/

[1] Avaya IP Office Platform Solution Description, Release 10.1, June 2017.
[2] Avaya IP Office Platform Feature Description, Release 10.1, Issue 1, June 2017.
[3] IP Office Platform 10.1 Deploying Avaya IP Office Platform IP500 V2, Document Number
15-601042, Issue 32f, 20 July 2017.
[4] Administering Avaya IP Office Platform with Manager, Release 10.1, July 2017.
[5] IP Office Platform 10.1 Using Avaya IP Office Platform System Status, Document 15-
601758, Issue 12d, 05 July, 2017.
[6] IP Office Platform 10.1 Using IP Office System Monitor, Document 15-601019, Issue 08d,
30 June, 2017.
[7] Using Avaya Communicator for Windows on IP Office, Release 10, August 2016.
[8] Administering Avaya Communicator on IP Office, Release 10.0, Issue 01.01, August 2016.

Additional Avaya IP Office documentation can be found at:

http://marketingtools.avaya.com/knowledgebase/

Product documentation for the Telmex SIP Trunking Service is available from Telmex.

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©2017 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.

Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at [email protected].

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