Avaya Algo 8028 SIP s Supplement

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Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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Avaya Solution & Interoperability Test Lab
Application Notes for Algo 8028 SIP Doorphone with Avaya
Aura® Session Manager and Avaya Aura® Communication
Manager - Issue 1.0
Abstract
These Application Notes describe the steps required to integrate the Algo 8028 SIP Doorphone
with Avaya Aura® Session Manager and Avaya Aura® Communication Manager configured
as an Evolution Server. The 8028 SIP Doorphone provides hands-free intercom capability and
entrance security with door unlock control. It is a SIP compliant device that registers with
Session Manager. The 8028 Doorphone includes a Control Unit and Door Station.
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 required to integrate the Algo 8028 SIP Doorphone
with Avaya Aura® Session Manager and Avaya Aura® Communication Manager configured as
an Evolution Server. The 8028 SIP Doorphone provides hands-free intercom capability and
entrance security with door unlock control. It is a SIP compliant device that registers with
Session Manager. The 8028 Doorphone includes a Control Unit and Door Station.
A visitor can press the call button on the Door Station to ring a specified telephone. The called
party can then answer the call to communicate with the Door Station. Using DTMF tones, the
called party can press a digit on the phone keypad to activate the door control relay to open the
door. Alternatively, a telephone can also originate a call to the Door Station, which would be
automatically answered. The 8028 Doorphone is configured via a web interface.
2. General Test Approach and Test Results
To verify interoperability of the 8028 Doorphone with Communication Manager and Session
Manager, calls were made from the doorphone to another specified telephone. The called
telephone would ring and answer the call. Upon answering the call, a two-way audio path was
established between the telephone and the Door Station. The called party would then be able to
press a digit on the telephone keypad to open the door. In addition, incoming calls to the
doorphone were also verified.
2.1. Interoperability Compliance Testing
Interoperability compliance testing covered the following features and functionality:
Successful registration of the 8028 SIP Doorphone with Session Manager.
Press call button at Door Station to ring specified telephone, answer the call, and
establish a two-way audio path. Caller ID on the telephone was also verified.
Called telephone can press a DTMF digit to open the door.
Incoming calls to the 8028 Doorphone.
G.711 codec support.
Proper system recovery after the 8028 Doorphone loses power.
2.2. Test Results
All test cases passed and the 8028 SIP Doorphone successfully registered with Session Manager.
Calls and delivery of DTMF tones to the doorphone were successful.
2.3. Support
For technical support on the 8028 SIP Doorphone, contact Algo Technical Support by phone,
through their website, or email.
Phone: [phone] (Canada & US only)
Web: http://www.algosolutions.com/support/support.html
Email: [e-mail]

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3. Reference Configuration
Figure 1 illustrates a sample configuration with an Avaya SIP-based network that includes
Session Manager, Communication Manager running on an Avaya S8800 Server with a G650
Media Gateway, and the Algo 8028 SIP Doorphone. Communication Manager was configured
as an Evolution Server and the 8028 SIP Doorphone registered with Session Manager.
Avaya Aura® Session Manager
(10.32.24.235)
LAN
Avaya G650 Media Gateway
Avaya 9600 Series SIP
and H. 323 Telephones
Avaya Aura® System Manager
Avaya S8800 Server running
Avaya Aura® Communication Manager
(C-LAN IP: 10.32.24.20)
SIP device registered
with Session Manager
Extension 77301 was used for
the door attendant telephone.
Algo 8028 SIP SIP Doorphone
(192.168.100.101)
SIP Extension: 78300
Figure 1: Avaya SIP Network with Algo 8028 SIP Doorphone

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4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Hardware Component
Version
Avaya S8800 Server and G650 Media
Gateway
Avaya Aura® Communication Manager 6.0.1
SP 3 (R016x.00.1.510.1 w/Patch 19009)
Avaya Aura® Session Manager
6.1 (6.1.2.0-612004)
Avaya Aura® System Manager
6.1 (6.1.0.07345-6.1.5.106)
Avaya 9600 Series IP Telephones
3.1 (H.323)
2.6.4 (SIP)
Algo 8028 SIP Doorphone
1.4
5. Configure Avaya Aura® Communication Manager
This section describes the steps for configuring the 8028 SIP Doorphone as an Off-PBX Station
(OPS) and configuring a SIP trunk between the Communication Manager and Session Manager.
Section 5.2 covers the station configuration for the 8028 SIP Doorphone. Use the System
Access Terminal (SAT) to configure Communication Manager and log in with the appropriate
credentials.
Note: If Communication Manager is already configured with a SIP trunk to Session Manager,
skip Section 5.1 and go directly to Section 5.2 to configure the station for the 8028 SIP
Doorphone.
5.1. Configure SIP Trunk
In the IP Node Names form, assign an IP address and host name for the S8800 Server processor,
the C-LAN board in the G650 Media Gateway, and Session Manager. The host names will be
used throughout the other configuration screens of Communication Manager.
change node-names ip Page 1 of 2
IP NODE NAMES
Name IP Address
Gateway001 10.32.24.1
ModMsg 192.50.10.45
clancrm 10.32.24.20
default 0.0.0.0
devcon-asm 10.32.24.235
medprocrm 10.32.24.21
procr 10.32.24.10
procr6 ::
( 8 of 8 administered node-names were displayed )
Use 'list node-names' command to see all the administered node-names
Use 'change node-names ip xxx' to change a node-name 'xxx' or add a node-name

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In the IP Network Region form, the Authoritative Domain field is configured to match the
domain name configured on Session Manager. In this configuration, the domain name is
avaya.com. By default, IP-IP Direct Audio (shuffling) is enabled to allow audio traffic to be
sent directly between IP endpoints without using media resources in the Avaya G650 Media
Gateway. The IP Network Region form also specifies the IP Codec Set to be used for calls
routed over the SIP trunk to Session Manager. This codec set is used when its corresponding
network region (i.e., IP Network Region „1‟) is specified in the SIP signaling group.
change ip-network-region 1 Page 1 of 20
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: avaya.com
Name:
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? y
UDP Port Max: 65535
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 34
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 7
Audio 802.1p Priority: 6
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
In the IP Codec Set form, select the audio codec type supported for calls routed over the SIP
trunk to the 8028 SIP Doorphone. The form is accessed via the change ip-codec-set 1
command. Note that IP codec set „1‟ was specified in IP Network Region „1‟ shown above. The
default settings of the IP Codec Set form are shown below. The 8028 SIP Doorphone supports
G.711.
change ip-codec-set 1 Page 1 of 2
IP Codec Set
Codec Set: 1
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.711MU n 2 20
2:
3:
4:
5:
6:
7:

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Prior to configuring a SIP trunk group for communication with Session Manager, a SIP signaling
group must be configured. Configure the signaling group form as follows:
Set the Group Type field to sip.
Set the IMS Enabled field to n.
The Transport Method field was set to tcp. In a production network, TLS transport
may also be used.
Specify the C-LAN board and the Session Manager as the two ends of the signaling
group in the Near-end Node Name field and the Far-end Node Name field,
respectively. These field values are taken from the IP Node Names form.
Ensure that the TCP port value of 5060 is configured in the Near-end Listen Port and
the Far-end Listen Port fields.
The preferred codec for the call will be selected from the IP codec set assigned to the IP
network region specified in the Far-end Network Region field.
Enter the domain name of Session Manager in the Far-end Domain field. In this
configuration, the domain name is avaya.com.
The Direct IP-IP Audio Connections field was enabled on this form.
The DTMF over IP field should be set to the default value of rtp-payload.
Communication Manager supports DTMF transmission using RFC 2833. The default
values for the other fields may be used.
add signaling-group 50 Page 1 of 1
SIGNALING GROUP
Group Number: 50 Group Type: sip
IMS Enabled? n Transport Method: tcp
Q-SIP? n SIP Enabled LSP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Near-end Node Name: clancrm Far-end Node Name: devcon-asm
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Secondary Node Name:
Far-end Domain: avaya.com
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? n Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

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Configure the Trunk Group form as shown below. This trunk group is used for calls to the SIP
Phones. Set the Group Type field to sip, set the Service Type field to tie, specify the signaling
group associated with this trunk group in the Signaling Group field, and specify the Number of
Members supported by this SIP trunk group. Configure the other fields in bold and accept the
default values for the remaining fields.
add trunk-group 50 Page 1 of 21
TRUNK GROUP
Group Number: 50 Group Type: sip CDR Reports: y
Group Name: To devcon-asm COR: 1 TN: 1 TAC: 1050
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 50
Number of Members: 10

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5.2. Configure Station
Use the add station command to add station for the 8028 SIP Doorphone. Use 9640SIP for the
Station Type. The Name field is optional. Use the default values for the other fields. The SIP
station can also be configured automatically by Session Manager as described in Section 6.7.
add station 78300 Page 1 of 6
STATION
Extension: 78300 Lock Messages? n BCC: 0
Type: 9640SIP Security Code: TN: 1
Port: IP Coverage Path 1: COR: 1
Name: 78300, Doorphone Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 19
Message Lamp Ext: 78300
Display Language: english Button Modules: 0
Survivable COR: internal
Survivable Trunk Dest? y IP SoftPhone? n
IP Video? n
Use the change off-pbx-telephone station-mapping command to map the Communication
Manager extension to the same extension on Session Manager. Enter the field values shown
below. For the sample configuration, the Trunk Selection field is set to aar so that AAR call
routing is used to route calls to Session Manager. AAR call routing configuration is not shown
in these Application Notes. The Config Set value can reference a set that has the default
settings.
change off-pbx-telephone station-mapping 78300 Page 1 of 3
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station Application Dial CC Phone Number Trunk Config Dual
Extension Prefix Selection Set Mode
78300 OPS - 78300 aar 1

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6. Configure Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. The procedures include
adding the following items:
SIP domain
Logical/physical Locations that can be occupied by SIP Entities
SIP Entities corresponding to Session Manager and Communication Manager
Entity Links, which define the SIP trunk parameters used by Session Manager when
routing calls to/from SIP Entities
Define Communication Manager as Administrable Entity (i.e., Managed Element)
Application Sequence
Add SIP User for the 8028 SIP Doorphone
Session Manager, corresponding to the Session Manager Server to be managed by
System Manager
Configuration of Session Manager is accomplished by accessing the browser-based GUI of
System Manager using the URL “https://<ip-address>/SMGR”, where <ip-address> is the IP
address of System Manager. Log in with the appropriate credentials. The initial screen is
displayed as shown below. The configuration in this section will be performed under Routing
and Session Manager listed within the Elements box.

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6.1. Specify SIP Domain
Add the SIP domain for which the communications infrastructure will be authoritative. Do this
by selecting Domains on the left and clicking the New button (not shown) on the right. The
following screen will then be shown. Fill in the following:
Name: The authoritative domain name (e.g., avaya.com)
Notes: Descriptive text (optional).
Click Commit.
Since the sample configuration does not deal with any other domains, no additional domains
need to be added.

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6.2. Add Locations
Locations can be used to identify logical and/or physical locations where SIP Entities reside for
purposes of bandwidth management. To add a location, select Locations on the left and click on
the New button (not shown) on the right. The following screen will then be shown. Fill in the
following:
Under General:
Name: A descriptive name.
Notes: Descriptive text (optional).
Under Location Pattern:
IP Address Pattern: A pattern used to logically identify the location.
Notes: Descriptive text (optional).
The screen below shows addition of the BR-DevConnect location, which includes the
Communication Manager and Session Manager. Click Commit to save the Location definition.

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6.3. Add SIP Entities
In the sample configuration, a SIP Entity is added for Session Manager and the C-LAN in the
G650 Media Gateway.
6.3.1. Session Manager
A SIP Entity must be added for Session Manager. To add a SIP Entity, select SIP Entities on
the left and click on the New button (not shown) on the right. The following screen is displayed.
Fill in the following:
Under General:
Name: A descriptive name.
FQDN or IP Address: IP address of the signaling interface on Session Manager.
Type: Select Session Manager.
Location: Select the location defined previously.
Time Zone: Time zone for this location.
Under Port, click Add, and then edit the fields in the resulting new row as shown below:
Port: Port number on which the system listens for SIP
requests.
Protocol: Transport protocol to be used to send SIP requests.
Default Domain The domain used for the enterprise (e.g.,
avaya.com).
Defaults can be used for the remaining fields. Click Commit to save each SIP Entity definition.

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6.3.2. Communication Manager
A SIP Entity must be added for the Communication Manager. To add a SIP Entity, select SIP
Entities on the left and click on the New button (not shown) on the right. The following screen
is displayed. Fill in the following:
Under General:
Name: A descriptive name.
FQDN or IP Address: IP address of the signaling interface (e.g., C-LAN board)
on the telephony system.
Type: Select CM.
Location: Select the location defined previously.
Time Zone: Time zone for this location.
Defaults may be used for the remaining fields. Click Commit to save the SIP Entity definition.

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6.4. Add Entity Link
The SIP trunk from Session Manager to Communication Manager is described by an Entity link.
To add an Entity Link, select Entity Links on the left and click on the New button (not shown)
on the right. Fill in the following fields in the new row that is displayed:
Name: A descriptive name (e.g., devcon13 Link).
SIP Entity 1: Select the Session Manager.
Protocol: Select the appropriate protocol.
Port: Port number to which the other system sends SIP
requests.
SIP Entity 2: Select the name of Communication Manager.
Port: Port number on which the other system receives
SIP requests.
Trusted: Check this box. Note: If this box is not checked,
calls from the associated SIP Entity specified in
Section 6.3.2 will be denied.
Click Commit to save the Entity Link definition.

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6.5. Define Communication Manager as Managed Element
Before adding SIP users, Communication Manager must be added to System Manager as a
managed element. This action allows System Manager to access Communication Manager over
its administration interface. Using this administration interface, System Manager will notify
Communication Manager when new SIP users are added.
To define Communication Manager as a managed element, select
ElementsInventoryManage Elements on the left and click on the New button (not shown)
on the right. In the New Entities Instance screen (not shown), select CM in the Type field, then
click Commit.
In the New CM Instance screen, fill in the following fields as follows:
In the Application tab:
Name: Enter an identifier for Communication Manager.
Type: Select CM from the drop-down field.
Node: Enter the IP address of the administration interface for
Communication Manager.

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In the Attributes tab:
Login / Password: Enter the login and password used for administration
access.
Is SSH Connection: Enable SSH access.
Port: Enter the port number for SSH administration access
(5022).
Defaults can be used for the remaining fields. Click Commit to save the settings.

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6.6. Add Application Sequence
To define an application for Communication Manager, navigate to Elements Session
Manager Application Configuration Applications on the left and select New button (not
shown) on the right. Fill in the following fields:
Name: Enter name for application.
SIP Entity: Select the Communication Manager SIP entity.
CM System for SIP Entity Select the Communication Manager managed element.
Click Commit to save the Application definition.

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Next, navigate to Elements Session Manager Application Configuration 
Application Sequences to define the Application Sequence for Communication Manager as
shown below. Provide a Name (e.g., DEVCON App Sequence) for the Application Sequence and
under Available Applications, click on the plus ( ) sign by DEVCON-APP to add it under the
Application in this sequence section.
Verify a new entry is added to the Applications in this Sequence table and the Mandatory
column is as shown below.
Note: The Application Sequence defined for Communication Manager Evolution Server can
only contain a single Application.

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6.7. Add SIP User
Add a SIP user for the 8028 SIP Doorphone.The following configuration will automatically
create the SIP station on Communication Manager Evolution Server.
To add new SIP users, navigate to Users User Management Manage Users from the left
and select New button (not shown) on the right.
Enter values for the following required attributes for a new SIP user in the Identity tab of the
new user form.
Last Name: Enter the last name of the user.
First Name: Enter the first name of the user.
Login Name: Enter <extension>@<sip domain> of the
user (e.g., [email protected]).
Authentication Type: Select Basic.
Password: Enter the password which will be used to
log into System Manager
Confirm Password: Re-enter the password from above.
The screen below shows the information when adding a new SIP user to the sample
configuration.