Avaya Definity G3

Configuration Notes

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Parlance is committed to the success of your operator services. Parlance’s Operator Assistant allows contact centers to meet

and exceed customer satisfaction goals while enhancing operator staff productivity. The flexible nature of Operator Assistant’s

architecture provides fast and easy implementation into a wide array of different PBX environments.

This document outlines the requirements and processes needed to ensure a successful Operator Assistant implementation with the

Avaya Definity G3.

Digital IntegrationSetup Instructions

1. Set up 4 8434D digital station ports with a Class of Service that provides:

➔ trunk-to-trunk transfer

(set in “system-parameters features” – 1st page OR the station must have a COS with

“trunk –to-trunk transfer override” set to “y”)

➔ 1 primary appearance, so that the phone can take only one call at a time

(On the 3rd page of the station form, there should only be 2 “call-appr” set on the

feature keys)

➔ automatic dialtone on pickup (usually system default)

➔ The COR of the 8434 must allow outcalling and not be restricted from connecting to other

station COR’s in the system

2. Put the stations in ucd (universal call distribution) or ddc (“direct department calling”) hunt group:

➔ The hunt group must have “n” for the following: acd, queue, vector

➔ lwc and messaging should be set to “none”

3. Set the coverage path for the hunt on busy/no answer to the switchboard or an auto attd, if desired

(in the event the Parlance ports are out-of-service)

4. To enable separate Dialed numbers for the system:

➔ Set up a coverage path to the hunt, with the first point set to “h#’ where “#” is the number

of the Parlance hunt group. [i.e. “h1” refers to hunt group 1.

➔ For each number, create a station with port = x, (or a hunt with no members), and set its

coverage path to the one you just set up for the Operator Assistant.

➔ Set the station’s name field to indicate the dialed number as the final part of the display

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Sample Station Settings for Operator Assistant Digital Port

General Settings

Extension: 1234

Type: 6408D+

Port: 01D5678

Name: Operator Assistant Line 1

Lock Messages? n

Security Code:

Coverage Path 1:

Coverage Path 2:

Hunt-to Station:

BCC: 0

TN: 1

COR: 6

COS: 1

Station Options

Loss Group: 2

Data Module? n

Speakerphone: 2-way

Display Language: english

Personalized Ringing Pattern: 1

Message Lamp Ext: 1234

Mute Button Enabled: y

Feature Options

LWC Reception: spe

LWC Activation? y

LWC Log External Calls? n

CDR Privacy? n

Redirect Notification? y

Per Button Ring Control? n

Bridged Call Alerting? n

Active Station Ringing: single

H.320 Conversion? n

AUDIX Name:

Messaging Server Name:

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Auto Select Any Idle Appearance? n

Coverage Msg Retrieval? y

Auto Answer: none

Data Restriction? n

Idle Appearance Preference? n

Per Station CPN – Send Calling Number? y

udible Message Waiting? n

Display Client Redirection? n

Select Last Used Appearance? n

Coverage After Forwarding? s

Considerations

Call Accounting

The call detail report (CDR) features of the G3, Revisions 4 and higher, provide information sufficient to associate toll calls with

the original caller when the Operator Assistant is used. Outgoing Trunk Call Splitting (OTCS) must be enabled to allow the proper

information to appear in the CDR output. The OTCS feature may be modified on page 1 of the CDR System Parameters form.

Device Type

The digital integration is via 8434 (2-wire) station emulation. However, in addition to “8434D” an

acceptable device type setting is “6408D+”.

Distinctive Audible Alert

The Distinctive Audible Alert station feature, which makes available the three ring types, should be set to “n” for all Operator As-

sistant lines.

Hunt Group

It is recommended that the hunt group be configured as UCD (Uniform Call Distribution). With settings of either “ucd-loa” (least

occupied agent) or “ucd-mia” (most idle agent), calls are evenly distributed among the Operator Assistant ports. Additionally,

overflow and no-answer forwarding points are also recommended. Using the switchboard as the overflow and no-answer for-

warding point allows callers to be serviced when either all lines are busy from usage or the Operator Assistant does not answer

due to a rare but possible system failure.

Terms

Call Detail Recording (CDR)

System logging that contains information about particular calls controlled by the switch. This information is useful for call cost-

ing, diagnostics, abuse detection, and network optimization.

Call Splitting

Keeps track of calls where more than two parties are involved. These can be calls that are transferred, conferenced, or where an

attendant becomes involved. If you have call splitting enabled and any of these situations arise, CDR produces a separate record

for each new party on the call.

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Class of Restriction (COR)

A COR is used to manage calling privileges through control of call origination and termination. Some facilities to which CORs can

be assigned are as follows: attendant console, hunt groups, stations, and trunk groups.

Terms

Call Detail Recording (CDR)

System logging that contains information about particular calls controlled by the switch. This information is useful for call cost-

ing, diagnostics, abuse detection, and network optimization.

Call Splitting

Keeps track of calls where more than two parties are involved. These can be calls that are transferred, conferenced, or where an

attendant becomes involved. If you have call splitting enabled and any of these situations arise, CDR produces a separate record

for each new party on the call.

Class of Restriction (COR)

A COR is used to manage calling privileges through control of call origination and termination. Some facilities to which CORs can

be assigned are as follows: attendant console, hunt groups, stations, and trunk groups.

Class of Service (COS)

Used to administer access permissions for call processing features that require dial code and/or feature button access permis-

sion.

Outgoing Trunk Call Splitting (OTCS)

If active, CDR starts a new record whenever an outgoing trunk call is involved in a conference or transfer. Whenever the call is

successfully transferred, CDR outputs the record relevant to this user’s participation. For conferenced calls, the originator of the

conference will be charged until he or she drops from the call, at which point CDR begins a second record for the conferenced

user. Records for parties on a conference do not overlap; they are split. These call records show the amount of time each party

was on the call, the outgoing trunk access code, the dialed number and the condition code, as well as the other fields specified in

the record format.

Terms courtesy of the Lucent Technologies DEFINITY Enterprise Communications Server Manual

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SIP ConfigurationThese Application Notes describe the steps for configuring Session Initiation Protocol (SIP) connectivity

between the Parlance Operator Assistant and an Avaya SIP solution including Communication

Manager, Session Manager, and various telephony endpoints. These endpoints include SIP, H.323,

analog and digital phones.

SIP is a standards-based communications approach designed to provide a common framework to

support multimedia communication. RFC 3261 is the primary specification governing this protocol. In

the configuration described in these Application Notes, SIP is used as the signaling protocol between

the Avaya components and the Parlance Operator Assistant server. SIP manages the establishment

and termination of connections as well as information such as codecs, calling party identity and calling

destination.

SIP Connectivity Configuration

Figure 1 below demonstrates the configuration used for this test environment. Although tested on

Midsize Business Template in the lab, for illustration purposes the Communication Manager and

Session Manager are logically identified. SIP trunks are built between the Avaya CM and SM, as well

as between SM and the Operator Assistant server. Testing was performed using a 5-digit dial plan

within the enterprise.

Figure 1

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Equipment and Software Validated

Equipment Software

Avaya S8800 Server Avaya Aura Midsize EnterpriseSolution Template 6.1 installed on SystemPlatform 6.0.3 • Communication Manager 6.0.1 • Session Manger 6.1 • System Manager 6.1

Parlance Server Windows Server

Avaya 9600 Series Telephone (H.323) Avaya one-X Deskphone 3.103S

Configure Avaya Communications Manager

This section covers the sample administration to allow SIP signaling between Avaya Communication Manager and Session

Manager using the Processor Ethernet interface via the SAT terminal.

Be sure to verify all of the appropriate licensing and features are present and enabled. This includes SIP trunk licenses, ARS, IP

Trunks, Trunk to Trunk Transfer, etc.

In the example configuration the system was configured with a 5-digit dial plan for local extensions, 9 for the ARS code, and *

or # for access codes. The lead number used to route calls to Operator Assistant is 39995. A shortened version of the dialplan

analysis screen is shown below.

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A Node-Name is needed to map an IP address to a name which can be used in various SAT screens. The following change node-

name screen illustrates the node-name for Session Manager and the local Processor Ethernet interface used in the SIP trunk

configuration.

Network Regions are used to logically group resources when deploying IP telephones, gateways and adjuncts. In this

configuration, Network Region 1 is used by Communication Manager and the IP telephones connected to it. Network Region

100 is used by the Session Manager and its connected components, including Operator Assistant. The two ip-network-region

screens are shown below to demonstrate the specific settings applicable to this environment.

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In addition to the configuration settings on the first screen, connectivity between the two regions must be established. The test

environment is configured with no bandwidth limit and utilizes the codec set 2. The ip-codec-set is shown below and utilizes both

G.729 and G.711MU as available codecs when establishing media communication.

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In order for SIP connectivity to the Session Manager to be in place, a signaling group and trunk group must be built. The signaling

type is SIP, and the two node-names previously configured will be used as the Near and Far end Node-Names. In this case, TLS was

used as the Transport Method being that two Avaya devices are communicating with each other. Also be sure to make note of the

two Listening Ports used by each of the nodes to be properly configured on both sides, the default 5061 is used here.

A trunk group, route pattern and lead number for Operator Assistant must also be administered in Communication Manager, but is

outside of the scope of this configuration document. There are many options such as AAR, UDP, virtual stations, etc. to achieve the

routing and numbering format sent to the Operator Assistant via Session Manager.

Configure System & Session Manager

This section demonstrates the needed configuration of the Avaya Aura Session Manager. This is accomplished via the System

Manager Web Interface. Once logged in with the proper credentials, most items are configured in Routing, located in the Elements

column.

First to be configured are the necessary SIP Entities. In the below screen shot, SIP Entities for both the Communication Manager at

172.27.1.42, and the Parlance Operator Assistant at 172.27.1.20 have been added to the System Manager configuration. The CM

server has been added with type CM and the Operator Assistant is configured as type Other, both using the appropriate IP address

for their server signaling interface.

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The next step is to configure the Entity Link between the Session Manager and Operator Assistant. For these configuration notes,

UDP signaling was used on port 5060. Select the two Entities from the appropriate drop-down menus and select the Connection

Policy as Trusted.

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Following the creation of the Entity Link, a Routing Policy must be added in order to route our lead number, 39995 to the Operator

Assistant server. Select Routing Policies on the menu and click to add a new Routing Policy, a screen shot is shown below. At this

point we will only give the Routing Policy a name and select the SIP Entity as Destination to be the Parlance Entity we created earlier.

Now a Dial Pattern must be assigned to the Routing Policy. In the first field, Pattern, we enter in the 5-digit lead number to Operator

Assistant, 39995. Our Min and Max values are set to be exactly the 5 digits we are expecting to receive, and the SIP Domain is set to

route for All Domains. Select Add under Originating Locations and Routing Policies in order to configure the destination for the Dial

Pattern. All applicable locations in the environment should be selected and the appropriate Routing Policy should be selected, in this

instance named toParlance.

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When communicating with Avaya Session Manager, the Parlance Operator Assistant utilizes the IP address of the Session Manager

when forming the URI in its SIP messages. In order to properly allow the routing of these SIP messages, the IP address must be

added in the Domains configuration screen in addition to the domain used when communicating with the Avaya Communication

Manager server. In the Domains screen shot below, both parlancecorp.com and 172.27.1.31 are domains which Session Manager

is aware of and able to route. They are added as type SIP.

Lastly, an Adaptation must be created and applied to the Parlance SIP Entity. From the Adaptation menu on the left, select New

to create a new Adaptation. In this configuration the Adaptation name was given ParlanceNameConnector and the Module name

used is the DigitConversionAdapter. In this Adaptation, our goal is to alter the domain portion of the SIP message’s URI. On the

Communication Manager segment of the communication, SIP messages are passed using ‘@parlancecorp.com’ while the Parlance

server exchanges SIP messages with ‘@172.27.1.31’ as the domain in the URI. To achieve compatibility on both sides we need to

alter the domains as the come in and out of the Session Manager. The following string was used to accomplish this:

ingressOverrideDestinationDomain=macsourceinc.com osrcd=172.27.1.31 fromto=true

This first instructs the incoming destination domain to be overwritten to the corporate domain of ‘macsourceinc.com’. Following

that is a single space, then the instruction to overwrite the outgoing source domain with the IP address of the Session Manager,

172.27.1.31, to allow for routing in the Parlance nameConnector. The last key/value pair indicates to perform this step on the From

and To headers in addition to the other headers within the SIP message that are changed by default.

parlancecorp.com

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In order for the Adaptation to be utilized, we must identify what matching dial patterns this should apply to. For incoming calls to SM,

we want to alter all calls either to external 9 plus 11-digit strings to route to ARS and out trunks, as well as 5-digit strings to internal

extensions. For calls destined for the Operator Assistant, we match on calls to the lead 39995 number as well as any external calls

again.

When this has been completed, return to the SIP Entity screen and assign the Adaptation.