Avaya ALGO-8028-SM User Manual

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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.

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

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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: (877) 884-2546 (Canada & US only) Web: http://www.algosolutions.com/support/support.html Email: support@algosolutions.com

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3. Reference Configuration

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 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.

Figure 1: Avaya SIP Network with Algo 8028 SIP Doorphone

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4. Equipment and Software Validated

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

The following equipment and software were used for the sample configuration provided:

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.