T-Library Functionality Support. Genesys 8.1

Chapter

6

T-Library Functionality

Support









This chapter describes the T-Library functionality that SIP Server supports. It contains the following sections:

Using T-Library Functions, page 413

Using the Extensions Attribute, page 422

Error Messages, page 437

Known Limitations, page 441

Using T-Library Functions

Table 102 presents the T-Library functionality supported in SIP Server. The

table entries use these notations:

N—Not supported

Y—Supported

E—Event only is supported

I—Supported, but reserved for Genesys Engineering

In Table 102

, when a set of events is sent in response to a single request, the events are listed in an arbitrary order. An asterisk (*) indicates the event that contains the same

Reference ID

as the request. For more information, refer to the Genesys Events and Models Reference Manual and the Platform SDK 8.x

.NET (or Java) API Reference for complete information on the T-Server events, call models, and requests.

Table 102 reflects only the switch functionality that Genesys software supports

and might not include the complete set of events that the switch offers.

Certain requests listed in

Table 102

are reserved for Genesys Engineering and are listed here merely for completeness of information.

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Chapter 6: T-Library Functionality Support Using T-Library Functions

Table 102: Supported Functionality

Feature Request

TOpenServer

TOpenServerEx

TCloseServer

TSetInputMask

TDispatch

TScanServer

TScanServerEx

TRegisterAddress a

TUnregisterAddress a

TMakeCall b

TAnswerCall

TReleaseCall

TClearCall

THoldCall

TRetrieveCall

TRedirectCall d

TMakePredictiveCall e

Request Subtype Corresponding Event(s) Supported

General Requests

EventServerConnected

EventServerConnected

EventServerDisconnected

EventACK

Y

Y

Y

Y

Y

Y

Y

Not Applicable

Not Applicable

Not Applicable

Registration Requests

EventRegistered

EventUnregistered

Call-Handling Requests

EventDialing Regular

DirectAgent

SupervisorAssist

Priority

DirectPriority

Y

Y

Y

N

N

N

N

Y c

Y

EventEstablished

EventReleased

EventReleased

EventHeld

EventRetrieved

EventReleased

EventDialing*

EventQueued

Y

Y

Y

Y

Y

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Table 102: Supported Functionality (Continued)

Feature Request

TInitiateTransfer b

TCompleteTransfer

TInitiateConference b

TCompleteConference

TDeleteFromConference

TReconnectCall

TAlternateCall

TMergeCalls

TMuteTransfer b,f

TSingleStepTransfer b

TSingleStepConference

Request Subtype Corresponding Event(s) Supported

Transfer/Conference Requests

Y EventHeld

EventDialing*

EventReleased*

EventPartyChanged

Y

Y EventHeld

EventDialing*

EventReleased*

EventRetrieved

EventPartyChanged

EventPartyAdded

EventPartyDeleted*

EventReleased

EventReleased

EventRetrieved*

Y

Y

Y

ForTransfer

ForConference

EventHeld*

EventRetrieved

EventHeld

EventReleased*

EventRetrieved

EventPartyChanged

EventHeld

EventReleased*

EventRetrieved

EventPartyChanged

EventPartyAdded

EventHeld

EventDialing*

EventReleased

EventPartyChanged

EventReleased*

EventPartyChanged

EventRinging*

EventEstablished

Y

N

N

Y

Y

Y

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Table 102: Supported Functionality (Continued)

Feature Request

TRouteCall b

TApplyTreatment

Request Subtype

Unknown

Default

Call-Routing Requests

EventRouteUsed

Label

OverwriteDNIS

DDD

IDDD

Direct

Reject

Announcement

PostFeature

DirectAgent

Priority

DirectPriority

AgentID

Corresponding Event(s) Supported

Y

Y

N

N

N

N

N

Y

N

N

N

N

N

N

N CallDisconnect

Call-Treatment Requests

Unknown

IVR

(EventTreatmentApplied +

EventTreatmentEnd)/

EventTreatmentNotApplied

Music

RingBack

N

N

Y

Y

Silence

Busy

CollectDigits

PlayAnnouncement

Y

Y

Y

Y

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Table 102: Supported Functionality (Continued)

Feature Request

TApplyTreatment

(continued)

Request Subtype

PlayAnnouncementAnd-

Digits

PlayApplication

VerifyDigits

RecordUserAnnouncement

DeleteUserAnnouncement

CancelCall

SetDefaultRoute

TextToSpeech

TextToSpeechAndDigits

FastBusy

Corresponding Event(s) Supported

TGiveMusicTreatment

TGiveRingBackTreatment

TCollectDigits

TSendDTMF

EventTreatmentApplied

EventTreatmentApplied

TGiveSilenceTreatment EventTreatmentApplied

DTMF (Dual-Tone Multi-Frequency) Requests

EventDigitsCollected

EventDTMFSent

Voice-Mail Requests

TOpenVoiceFile

TCloseVoiceFile

TLoginMailBox

TLogoutMailBox

TPlayVoice

TAgentLogin

TAgentLogout

EventVoiceFileOpened

EventVoiceFileClosed

EventMailBoxLogin

EventMailBoxLogout

EventVoiceFileEndPlay

Agent & DN Feature Requests

EventAgentLogin

EventAgentLogout

Y

N

Y

N

N

N

N

N

Y

Y

N

N

N

N

Y

N

Y

N

N

N

Y

N

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Chapter 6: T-Library Functionality Support Using T-Library Functions

Table 102: Supported Functionality (Continued)

Feature Request

TAgentSetIdleReason

TAgentSetReady

TAgentSetNotReady

TMonitorNextCall

TCancelMonitoring

TCallSetForward

TCallCancelForward

TSetMuteOff g

TSetMuteOn

g

TListenDisconnect

TListenReconnect

TSetDNDOn

TSetDNDOff

TSetMessageWaitingOn

TSetMessageWaitingOff

TQuerySwitch a

OneCall

AllCalls

None

Request Subtype

Unconditional

OnBusy

OnNoAnswer

OnBusyAndNoAnswer

SendAllCalls

DateTime

ClassifierStat

Corresponding Event(s) Supported

EventAgentIdleReasonSet N

EventAgentReady

EventAgentNotReady

EventMonitoringNextCall

Y

Y

Y

Y

EventMonitoringCanceled

EventForwardSet

N

N

N

Y

Y

N

Y

Y

EventForwardCancel

EventMuteOff

EventMuteOn

EventListenDisconnected

EventListenReconnected

EventDNDOn

EventDNDOff

EventMessageWaitingOn

EventMessageWaitingOff

Query Requests

EventSwitchInfo

N

N

Y

Y

Y

Y

Y

Y

N

N

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Table 102: Supported Functionality (Continued)

Feature Request

TQueryCall a

TQueryAddress a

Request Subtype

PartiesQuery

StatusQuery

AddressStatus

MessageWaitingStatus

AssociationStatus

CallForwardingStatus

AgentStatus

NumberOfAgentsInQueue

NumberOfAvailableAgents-

InQueue

Number Of Calls InQueue

AddressType

CallsQuery

SendAllCallsStatus

QueueLoginAudit

NumberOfIdleClassifiers

NumberOfClassifiersInUse

NumberOfIdleTrunks

NumberOfTrunksInUse

DatabaseValue

DNStatus

QueueStatus

Corresponding Event(s) Supported

EventPartyInfo N

EventAddressInfo

N

N

Y

Y

Y

Y

N

Y

N

N

N

Y

Y

Y

Y

N

Y

N

N

Y

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Table 102: Supported Functionality (Continued)

Feature Request

TQueryLocation

TQueryServer a a

Request Subtype

AllLocations

LocationData

MonitorLocation

CancelMonitorLocation

MonitorAllLocations

Corresponding Event(s) Supported

EventLocationInfo

CancelMonitorAllLocations

EventServerInfo

User-Data Requests

h

I

I

I

I

I

I

Y

Y

Y

Y

Y

TAttachUserData EventAttachedDataChanged

TUpdateUserData

TDeleteUserData

TDeleteAllUserData

EventAttachedDataChanged

EventAttachedDataChanged

EventAttachedDataChanged

ISCC (Inter Server Call Control) Requests

TGetAccessNumber b

TCancelRegGetAccess-

Number

EventAnswerAccessNumber

EventReqGetAccess-

NumberCanceled

Special Requests

TReserveAgent

TSendEvent

TSendEventEx

TSetCallAttributes

TSendUserEvent

TPrivateService

TNetworkConsult

TNetworkAlternate

EventAgentReserved

EventACK

EventACK

EventCallInfoChanged

EventACK

EventPrivateInfo

Network Attended Transfer/Conference Requests i

EventNetworkCallStatus

EventNetworkCallStatus

I

I

Y

Y

I

I

I

I

Y

Y

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Table 102: Supported Functionality (Continued)

Feature Request

TNetworkTransfer

TNetworkMerge

TNetworkReconnect

TNetworkSingleStep-

Transfer

TNetworkPrivateService

Request Subtype

TTransactionMonitoring

Corresponding Event(s) Supported

EventNetworkCallStatus Y

EventNetworkCallStatus

EventNetworkCallStatus

EventNetworkCallStatus

N

Y

N

EventNetworkPrivateInfo

ISCC Transaction Monitoring Requests

EventACK

EventTransactionStatus

N

Y

E a. Only the requestor receives a notification of the event associated with this request. b. This feature request may be made across locations in a multi-site environment. However, if the location attribute of the request contains a value relating to any location other than the local site—except when the response to this request is

EventError

—there will be a second event response that contains the same reference

ID

as the first event. This second event will be either

EventRemoteConnectionSuccess

or

EventRemoteConnectionFailed.

c. Supported for SIP endpoints that have the Remote Talk feature activated.

d. SIP Server treats unsuccessful redirect operations differently, depending on whether they were initiated through 3pcc (

TRedirectCall

) or 1pcc (

302 Moved

). Incase of

TRedirectCall

, if the destination rejects the redirect for some reason (capacity, DND, for example), the call remains alive. In case of an unsuccessful SIP

302 Moved

attempt, the call is rejected. The following sample scenario explains the difference:

- DN1 has reached its configured capacity.

- DN2 is dialing DN3, and DN3 is ringing.

- DN 3 redirects to DN1.

In case of a 3pcc redirect, the call between DN2-DN3 will remain in the ringing state. In case of a 1pcc redirect, the call between DN2-DN3 is released.

e. SIP Server does not use the extensions

parameter. Any data in this parameter is ignored.

f. SIP Server supports

TMuteTransfer

requests with the following limitations:

- The

TMuteTransfer

operation does not support the out-of-signaling-path (OOSP) transfer mode.

- The

TMuteTransfer

operation to a Hunt Group will be completed only after a Hunt Group member picks up the call.

g. SIP Server supports

TSetMuteOn

and

TSetMuteOff

only for established conferences, to allow for service observing h. Two subtypes are supported by

EventLocationInfo

:

LocationInfoLocationMonitorCanceled

and

LocationInfoAllLocationsMonitorCanceled.

i. All T-Servers support NAT/C requests with

AttributeHomeLocation

provided that this attribute identifies a network location that is capable of processing such requests. Refer to the Network T-Server Deployment

Guides to determine whether a specific Network T-Server can process these requests.

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Chapter 6: T-Library Functionality Support Using the Extensions Attribute

Using the Extensions Attribute

SIP Server supports the use of the

Extensions

attribute as documented in the

Genesys Events and Models Reference Manual and the Platform SDK 8.x .NET

(or Java) API Reference. See those documents for complete information on the

T-Server events, call models, and requests. Additionally, the

Extensions described in

Table 103 are also supported.

Table 103: Use of the Extensions Attribute

Extension Used In

Key Type

AOC-Destination-DN after-call-divertdestination

AnsMachine

FaxDest

String

String

String

String

Description

Advice of Charge

TPrivateService If set, the value of this key points to an existing party in the call in the established state.

See “Providing AoC Notifications for

Established Calls” on page 104 for details.

Call Divert Destination

TRouteCall A destination DN where SIP Server will divert the call in cases where the caller remains on the line when all other parties have left.

See “Call Divert Destination” on page 121

for details.

Call Progress Analysis

TMakePredictive-

Call

TMakePredictive-

Call

If set, the value of this key (can only be set to drop

) overrides the Application-level parameter

am-detected

. If the CPD result shows that the predictive call reached an answering machine,

SIP Server drops the call.

See “Outbound IP Solution Integration” on page 317 for details.

If set, the value of this key (can only be set to drop

) overrides the Application-level parameter

fax-detected

. If the CPD result shows that the

predictive call reached a fax machine, SIP

Server drops the call.

See “Outbound IP Solution Integration” on page 317 for details.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key

SilenceDest

Type

String record String

TMakePredictive-

Call

If set, the value of this key (can only be set to drop

) overrides the Application-level parameter

silence-detected

. If the CPD result shows that silence is detected, SIP Server drops the call.

See “Outbound IP Solution Integration” on page 317 for details.

Call Recording

TRouteCall,

TPrivateService

For

TRouteCall

:

• When set to destination, call recording is initiated on the routing destination DN

(agent), and will continue until the agent leaves the call.

• When set to source, call recording is initiated on the DN that sent a call to the

Routing Point (customer), and will continue until the customer leaves the call.

See “Call Recording—NETANN-Based” on page 126

for details.

To disable call recording, in the routing strategy, configure the

TRouteCall

request to include the record

key with the appropriate value, as follows:

•

disable_source

—to disable recording on the origination DN.

• disable_destination

—to disable recording on the destination DN.

See “DN Recording Override” on page 137 for details.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key

record

(cont.)

Type

MonitorMode

MonitorScope

String

String

For

TPrivateService:

• When set to source,

call recording is initiated for

ThisDN

(the source DN for the request).

• When set to destination,

call recording is initiated for

OtherDN

(identified in the request).

See “Call Recording—MSML-Based” on page 130

for details.

Call Supervision

TMonitorNextCall,

TRouteCall,

TSetMuteOn,

TSetMuteOff,

EventPrivateInfo

Specifies the monitoring mode as follows:

• mute, normal

—A mute connection.

• connect

—A three-party conference call

(open supervision).

• coach

—Only the agent can hear the supervisor (whisper coaching).

If

MonitorMode

is set to coach

in the

TSetMuteOff

or

TSetMuteOn

request, the monitoring mode is changed to whisper coaching for the current supervision session.

Note:

TSetMuteOn

and

TSetMuteOff

support only the coach

value.

See “Call Supervision” on page 143 for details.

TMonitorNextCall,

TRouteCall

Specifies the required intrusion/observation scope. Values:

• agent

—The monitoring is initiated for a specific agent. The supervisor is disconnected when the call is transferred or released, but will be connected to the next call that is routed to the same agent.

• call

—The monitoring is initiated to track an entire customer call. If the call is transferred to another agent, queue, or VRU, the monitoring function continues with the call until the customer disconnects the call.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key

AssistMode

Type

String

Transfer-Type String original-dialplan-digits Integer

TSingleStep-

Conference

Specifies the required assistance mode. Values:

• connect

—This is the default value - a threeparty conference call.

• coach

—Only the agent can hear the supervisor (whisper coaching).

Call Transfer

TRouteCall the SIP call flow that will be used to deliver the call to a routing destination. Valid values:

• invite

—An

INVITE

transaction is used to connect an origination party with a routing destination.

• refer

—A

REFER

method is used to complete the call routing. The

Host

name in the

Refer-

To

header points to SIP Server. Therefore,

SIP Server remains in the signaling path when two parties are connected after the routing.

Note: The

REFER

method cannot be used if the call is not answered by the remote party, and no treatment is applied to the call.

• oosp

(Out Of Signaling Path)—SIP Server is taking itself out of the signaling path by sending the

REFER

request or a

302

response, pointing to the routing destination, to the origination party. Two parties are connected directly when routing is complete. SIP Server no longer controls the call.

See “Selecting SIP Call Flows from the Routing

Strategy” on page 169 for details.

Dial Plan

EventQueued,

EventRouteRequest,

EventRinging

If set, this key specifies the call’s original destination (dialed) number before the dial plan is applied to the call.

See “Dial Plan” on page 210 for details.

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Table 103: Use of the Extensions Attribute (Continued)

Key

DNIS_OVER

Extension

Type

Integer

Used In Description

UseDialPlan String

EventQueued,

EventRouteRequest

TRouteCall

Contains overdialed digits removed from DNIS when the dnis-max-length

dial-plan rule parameter is specified.

See “Removal Overdialed Digits From DNIS” on page 213 for details.

If set, specifies how SIP Server applies the dial plan:

• full

—The dial plan is applied to the destination of

TRouteCall

, including the digit translation and forwarding rules.

• partial

—Valid for both SIP Server and SIP

Feature Server dial plans. Only the digit translation is applied to a dial-plan target.

Forwarding rules, such as forwarding on no answer ( ontimeout

), forwarding on busy

( onbusy

), forwarding on DND ( ondnd

), forwarding on no response ( onunreach

), and forwarding on not SIP registered ( onnotreg

) are not applied.

• false

—No dial plan is applied to the destination of

TRouteCall

.

Notes:

• For ISCC calls, this extension is applied only to calls routed through an External Routing

Point (c ast-type=route-notoken).

• This extension is not supported in Business

Continuity deployments.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

divert-on-ringing sdp-c-host sdp-m-port-low

String

String

Integer

Divert On Ringing

TRouteCall Valid values:

• true

—SIP Server generates

EventRouteUsed and

EventDiverted

messages when any SIP

18x

response (

180 Ringing

or

183 Session

Progress

) arrives for the

INVITE

request at the routing destination.

• false

—SIP Server postpones .

EventRouteUsed

and

EventDiverted messages until the call is answered by the routing destination with a SIP

200 OK message. If the call is not answered within the value specified by the after-routingtimeout

option, the destination SIP dialog is canceled and an

EventError

message is generated.

Dummy SDP

TRouteCall The value will be propagated as the connection address in the c= line of Dummy SDP. Typically, this would be the IP address of the media server host.

See “Dummy SDP” on page 238 for details.

TRouteCall Any integer that represents a valid UDP port.

This value represents the low part in the range of ports to be included in the m=

line of the SDP.

Key rules:

• The actual port used can equal this low value.

• Only even-numbered ports are used.

See “Dummy SDP” on page 238 for details.

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Table 103: Use of the Extensions Attribute (Continued)

Extension

Key

sdp-m-port-high

Type

Integer

Used In Description

charge-type Integer

AgentLogoutOnUnregist er

String

TRouteCall Any integer that represents a valid UDP port.

This value represents the high part in the range of ports to be included in the m=

line of the SDP.

Key rules:

• The actual port used can equal the low value, but cannot equal this high value.

• Only even-numbered ports are used. For example, if the range is 4000 to 4010, the port in the

INVITE

can be

4000, 4002, 4004,

4006, and 4008

only.

• If you leave this parameter unspecified

(empty), then the value of the sdp-m-portlow will be used.

See “Dummy SDP” on page 238 for details.

Early Media

TApplyTreatment If set, the value of this key overrides the value set in the

charge-type

configuration option for the current call.

1

—Free. When SIP Server receives this key in the initial TApplyTreatment request, SIP Server forces audio treatments to be played in early media, free of charge, instead of media in the established state, in deployments where the

charge-type

option is set to

2

(charged).

Consecutive audio treatments are played in early media until the new

TApplyTreatment

request containing the charge-type

key set to

2

(charged) arrives.

2

—Charged. SIP Server forces audio treatments to be played in the established state and ignores the charge-type key value in consecutive

TApplyTreatment

requests.

See “Controlling Early Media with a Routing

Strategy” on page 247 for details.

Emulated Agents

TAgentLogin If set to true,

SIP Server logs out emulated agents on unregister.

See “Emulated Agents Support” on page 250

.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

agent-greeting customer-greeting agent-greeting-type

record-agent-greeting music music-on-hold

String

String

String

String

String

String

Greetings

TRouteCall

TRouteCall

TRouteCall

Specifies the name of the media file that will be used as a greeting for the agent.

See “Personal Greetings” on page 330

for details.

Specifies the name of the media file that will be used as a greeting for the customer.

See “Personal Greetings” on page 330

for details.

When set to vxml

, enables VXML support for

agent greeting. See “VXML Support for Agent

Greeting” on page 332 for details.

TRouteCall Specifies whether the agent greeting or customer greeting must be recorded when both recording and greeting are enabled for the call.

• If set to true,

the agent greeting is recorded.

• If set to false,

the customer greeting is recorded.

If set, the key value takes precedence over the

record-agent-greeting

option value.

See “Recording an Agent Greeting” on page 335

for details.

Music on Hold

THoldCall,

TAlternateCall,

TInitiateTransfer,

TInitiate-

Conference

TRouteCall

Specifies the file name for music on hold for the call.

See “Customizing Music on Hold and in Queue” on page 194 for details.

Specifies the name of the file that is played for the music-on-hold treatment when one of the parties in the call is placed on hold.

See “Customizing Music on Hold and in Queue” on page 194 for details.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

connect-nailedup-onlogin

ReasonCode

String

String

Nailed-up Connections

TAgentLogin

EventAgentNot-

Ready

If set, the value of this key overrides any value set in the

connect-nailedup-on-login

option

setting but only for a current login session.

Specifies SIP Server actions when receiving a

TAgentLogin

request from a DN with the configured nailed-up connection, as follows:

• When this key is set to a Routing Point number, SIP Server immediately establishes a nailed-up connection between an agent’s endpoint and the specified Routing Point.

After processing the

TRouteCall

request to the gcti::park device, SIP Server parks the agent on gcti::park

, establishing the persistent SIP connection with the agent’s endpoint.

• When this key is set to gcti::park

, SIP

Server parks the agent on the gcti::park device directly, establishing the persistent

SIP connection with the agent’s endpoint.

• When this key is set to an empty value, SIP

Server disables this feature for a particular agent in a current login session.

See “Nailed-Up Connections” on page 298

for details.

Value:

NailedUpConnectionTerminated

Specifies that the nailed-up connection is terminated.

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Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

AFTER_ROUTING_TIMEO

UT_ACTION after-routing-timeout

NO_ANSWER_ACTION

String

Integer

String

No Answer Supervision

TRouteCall

TRouteCall

If set, the value of this key overrides any value set in the after-routing-timeout-action configuration option for the current call.

Values:

• none

—SIP Server takes no action.

• notready

—When an agent is logged in to a routing destination that does not answer the call, SIP Server sets this agent to

NotReady state.

• logout

—When an agent is logged in to a routing destination that does not answer the call, SIP Server logs this agent out.

See “No-Answer Supervision” on page 313 for

details.

A positive integer that overrides the value of the

after-routing-timeout

configuration option,

which specifies the length of time (in seconds) that SIP Server waits before diverting the call from the

Routing Point

DN to the destination

DN after

TRouteCall

was processed. When the call is not diverted before the timeout expires,

SIP Server generates an

EventError

message.

If set to a value of

0

(zero), this extension is ignored and the value of the after-routingtimeout

configuration option is used instead.

TRouteCall If set, the value of this key overrides any value set in any of the following configuration options for the current call:

• no-answer-action

• agent-no-answer-action

See “No-Answer Supervision” on page 313 for

details.

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Chapter 6: T-Library Functionality Support Using the Extensions Attribute

Table 103: Use of the Extensions Attribute (Continued)

Key

Extension

NO_ANSWER_OVERFLOW

NO_ANSWER_TIMEOUT

Type

Commaseparated list

Used In Description

ReasonCode

LCTPartiesLength

LCTParty<n> no-answer EventAgentNot-

Ready

If configured, reports

ReasonCode

set to noanswer

when an agent is placed in the

Not Ready state after not answering a call.

See “Reporting ReasonCode” on page 314

for details.

Providing Call Participant Info

(See “Providing Call Participant Info” on page 349 for details)

Integer EventUserEvent

Integer EventUserEvent

Specifies how many parties are involved in a single call.

Represents a party of the call, where n is an integer value starting from

0

.

LCTParty<n>_location String

LCTSupervisor<n>

String

Integer

TRouteCall

TRouteCall

EventUserEvent

EventUserEvent

If set, the value of this key overrides any value set in any of the following configuration options for the current call:

• no-answer-overflow

• agent-no-answer-overflow

• extn-no-answer-overflow

• posn-no-answer-overflow

See “No-Answer Supervision” on page 313 for

details.

If set, the value of this key overrides any value set in any of the following configuration options for the current call:

• no-answer-timeout

• agent-no-answer-timeout

• extn-no-answer-timeout

• posn-no-answer-timeout

See “No-Answer Supervision” on page 313 for

details

Provides the DN location name; that is, the name of the switch to which this DN belongs.

Represents the supervisor of the call, where n is an integer value starting from

0

.

432 Framework 8.1

Chapter 6: T-Library Functionality Support Using the Extensions Attribute

Table 103: Use of the Extensions Attribute (Continued)

Extension

Key

LCTSupervisor<n>_ location

Type

String

LCTSupervisor<n>_ monitoredDN

LCTSupervisor<n>_ mode

Integer

String

Used In

EventUserEvent

EventUserEvent

EventUserEvent

Description

Provides the name of the switch to which this supervisor belongs.

Represents the agent monitored by this supervisor.

Provides Supervision mode.

CPNDigits

DisplayName

String

String call_timeguard_timeout Integer

Providing Caller ID

TMakeCall,

TMakePredictive-

Call,

TInitiate-

Conference,

TInitiateTransfer,

TRouteCall,

TSingleStep-

Transfer,

TSingleStep-

Conference

If set, the value of this Extension overrides the username provided in the URI in the

From

header of the

INVITE

message.

This Extension is not applicable when performing a

TMakeCall

request using the

REFER method.

See “Providing a Caller ID” on page 349 for

details.

TMakeCall,

TMakePredictive-

Call,

TInitiate-

Conference,

TInitiateTransfer,

TRouteCall,

TSingleStep-

Transfer,

TSingleStep-

Conference

TMakePredictive-

Call

Allows users to define their own display name and Calling Party Number (CPN) for OCS outbound dialing

If set, the value of this Extension is used as the display name, in addition to the CPN digits, when providing the Caller ID. This setting only applies if the

CPNDigits

key-value pair is also present in the Extensions.

SIP Server maps this info from the request to the

From

header of the SIP

INVITE

. For example,

From: “DisplayName”

<sips:[email protected]>;tag=a48s

If set, the value of this key specifies the maximum time, in milliseconds, allowed for post-connect CPD results to be received from the device performing CPD (either Genesys Media

Server or Media Gateway). See “Outbound IP

Solution Integration” on page 317

for details.

SIP Server—Deployment Guide 433

Chapter 6: T-Library Functionality Support Using the Extensions Attribute

Table 103: Use of the Extensions Attribute (Continued)

Key

feature dn agent-dn login-id monitor-type password

Extension

post-feature-dn

Used In

Type

Remote Supervision

String

“remoteobserving”

TRouteCall

String

String

TRouteCall

TRouteCall

This extension triggers registration of a routed party as a supervisor with parameters specified in additional

Extensions

described in this subsection.

See “Remote Supervision” on page 158

for details.

An optional DN number that can be used during the monitoring session, as a substitute for the external PSTN number that the supervisor used to dial in.

The target that the supervisor wants to monitor.

This parameter can be a Routing Point, ACD

Queue, or an agent DN.

String TRouteCall

Description

AllCalls

String

String

TRouteCall

TRouteCall

TRouteCall

(Optional) A Login ID, which will be used for remote client authorization.

(Optional) The supervisor will monitor all consecutive calls for the selected target, until the supervisor decides to hang up and end the monitoring session. In between monitored calls, the supervisor’s call is parked.

(Optional) A password, which will be used for remote client authorization for a specified Login

ID.

(Optional) A Routing Point, to which the remote supervisor will be connected after the supervision session.

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Chapter 6: T-Library Functionality Support Using the Extensions Attribute

Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

geo-location overflow-location busy-on-reject

ConvertOtherDN

String

String

String

( true

, false

)

String

Routing

TRouteCall,

TApplyTreatment

TRouteCall,

TApplyTreatment

If

TRouteCall

or

TApplyTreatment

contains the geo-location extension, SIP Server makes this geo-location to be the most preferable on the current call. It means that each time when the switch resource is selected based on the geolocation parameter, resources with the preferred geo-location take precedence.

If the preferable geo-location is activated on the call, but

TRouteCall

or

TApplyTreatment requests does not contain the geo-location extension specified, SIP Server deactivates the preferred geo-location mode, and switch resources are selected in accordance with a regular selection procedure.

Note: The overflow-location

extension key applies only if the geo-location

extension key is defined in the same request.

See “Geo-Location for MSML-Based Services:

Strict Matching” on page 401 .

TRouteCall If SIP Server receives

TRouteCall

with busy-onreject=true

in

AttributeExtensions

and if the destination responds with the

486 Busy Here message, it generates

EventDestinationBusy

for the origination party.

Smart OtherDN Handling

See “Smart

OtherDN Handling” on page 371

.

A value of

0

disables all conversions for the call.

A value of

1

forces the relevant conversion for the call.

SIP Server—Deployment Guide 435

Chapter 6: T-Library Functionality Support Using the Extensions Attribute

Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

User-Agent

Server and User-Agent Headers

String TMakeCall,

TMakePredictive-

Call,

TSingleStep-

Transfer,

TSingleStep-

Conference,

TInitiateTransfer,

TInitiate-

Conference

Enables inserting the

User-Agent

header.

See “Enabling Server and User-Agent Headers” on page 200 .

Sending Outgoing INVITEs with Multipart Body

SIP_MIME_HEADERS String

Geolocation String

TRouteCall Value:

Geolocation:application/pidf+xml

Passes geolocation content from

TRouteCall

into an outgoing

INVITE

message.

See “Sending Outgoing INVITEs with Multipart

Body” on page 357 .

TRouteCall included into an outgoing

INVITE

message.

See “Sending Outgoing INVITEs with Multipart

Body” on page 357 .

Trunk Capacity

Dest-Capacity sdn-licenses-in-use Integer sdn-licenses-available

Integer TRouteCall Specifies the trunk capacity as defined in the

URS routing strategy. SIP Server checks the current load of the targeted Trunk, and if the

Dest-Capacity value is exceeded, SIP Server replies to the RouteRequest with a

403

Forbidden

error message.

T-Server Common Part Extensions

Integer

EventServerInfo Specifies how many SDN licenses are currently in use.

Specifies how many SDN licenses are currently available.

436 Framework 8.1

Chapter 6: T-Library Functionality Support Error Messages

Table 103: Use of the Extensions Attribute (Continued)

Extension Used In Description

Key Type

VideoFile

AgentVideo

String

String

Video Support

TSingleStep-

Conference

A string that contains the name of the video file that will be played for the customer. If this keyvalue pair is not specified, the default video file will be played. The default video file is configured in the SIP Server

Application

using the

default-video-file

configuration option.

See “Video Support” on page 390 for details.

TSingleStep-

Conference

A string that identifies the origin of the video stream played to the agent. The values are as follows:

• from-third-party

—The agent receives video from a third party—that is, the party that participated in the call before the operation started.

• to-third-party

—The agent receives the same video stream played to the third party—that is, video from the file specified by the

VideoFile

parameter.

See “Video Support” on page 390 for details.

Error Messages

Table 104 presents the complete set of error messages SIP Server distributes in

EventError,

which SIP Server generates when it cannot execute a request because of an error condition

.

Table 104: Error Messages for SIP Server

Code Symbolic Name

40 TERR_NOMORE_LICENSE

Description

No more licenses are available.

41 TERR_NOT_REGISTERED Client has not registered for the DN.

42 TERR_RESOURCE_SEIZED Resource is already seized.

43 TERR_IN_SAME_STATE Object is already in requested state.

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Chapter 6: T-Library Functionality Support Error Messages

Table 104: Error Messages for SIP Server (Continued)

Code

50

Symbolic Name

TERR_UNKNOWN_ERROR

51 TERR_UNSUP_OPER

52 TERR_INTERNAL

53 TERR_INVALID_ATTR

54 TERR_NO_SWITCH

55 TERR_PROTO_VERS

56 TERR_INV_CONNID

57 TERR_TIMEOUT

Description

Unknown error code. Request cannot be processed.

Operation is not supported.

Internal error.

Attribute in request operation is invalid.

No connection to the switch.

Incorrect protocol version.

Connection ID in request is invalid.

Switch or T-Server did not respond in time.

58

59

TERR_OUT_OF_SERVICE

TERR_NOT_CONFIGURED

Switch or T-Server is out of service.

DN is not configured in the Configuration

Database.

DN in request is invalid.

61 TERR_INV_CALL_DN

71 TERR_INV_CALD_DN Invalid called DN.

93 state.

96 TERR_CANT_COMPLETE_CONF

118 TERR_SERV_UNAVAIL

119 TERR_BAD_PASSWD

122

123

128

166

TERR_CANT_REG_DNS

TERR_DN_NOT_EXIST

TERR_BAD_DN_TYPE

TERR_RES_UNAVAIL

168 TERR_INV_ORIG_ADDR

173 TERR_UNSUCC_CONFER

177 TERR_TARG_DN_INV

Call cannot add new conference party.

Requested service is unavailable.

Password was invalid. May occur when SIP

Server receives a

404 Not Found message from the Media Gateway after an unsuccessful attempt to route a call.

Cannot register DNs on the switch.

DN for association does not exist.

Invalid DN type for DN registration.

(JTAPI object) resource is not available.

Originating address in request was invalid.

Unsuccessful conference.

DN target (in route call) was invalid.

438 Framework 8.1

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Table 104: Error Messages for SIP Server (Continued)

Code Symbolic Name

185 TERR_SET_WRONG_STATE

192 TERR_AGENT_ID_INV

195 TERR_CFW_DN_INV

212 TERR_ADMIN_DEV_DIS

223 TERR_BAD_PARAM

226 TERR_OUT_OF_MEM

231 TERR_DN_BUSY

232 TERR_DN_NO_ANSWER

233 TERR_CALL_REJECTED

236 TERR_TIMEOUT_PRF_OP

237 TERR_DISCON_CALL

243 TERR_CLNT_NOT_MON

259 TERR_INV_PASSWD

Description

Set is in the wrong state for invocation.

Agent ID is invalid.

Call forwarding address is invalid.

Device disabled by administration.

Bad parameter is passed to function.

Out of memory (local).

DN is busy.

No answer at a DN.

Call has been rejected.

Timeout performing operation.

Call has disconnected.

Internal error—client corrupted in T-Server.

Invalid credentials (login_id or password).

282 TERR_NO_VCHAN_AVAIL

291 TERR_OTHER_TEL_OPER

302 TERR_INV_DTMF_STRING

355 TERR_OPER_INVALID

No voice channel available.

Other telephony operation is in progress.

DTMF string invalid.

Operations are invalid on system messages.

410 TERR_INAPPR_TRTM Invalid treatment type.

415 DN in the request is invalid.

470

496

TERR_PARTY_NOT_ON_CALL

TERR_INV_CALL_STATE

506 TERR_RECVD_INV_STATE

527 TERR_ALRDY_SIGN_IN

549 TERR_OTHER_INV_FRMT

565 TERR_INVALID_STATE

Party in request is not involved in a call.

Party in request is in the call

state.

Call/Party is in invalid state for this time

The agent’s sign-in number is already active at another console.

Other invalid format.

Invalid state.

SIP Server—Deployment Guide 439

Chapter 6: T-Library Functionality Support Error Messages

Table 104: Error Messages for SIP Server (Continued)

Code Symbolic Name

649 TERR_ASAI_OUT_CALL_BARRED

700 TERR_INV_LOGIN_REQ

701 TERR_INV_LOGOUT_REQUEST

702 TERR_INV_READY_REQ

726 TERR_DMS_INV_AUTHCODE

1141 TERR_CSTA_OPER_REQ_INCOMPAT

1143 TERR_CSTA_OPER_OBJ_NOT_KNOWN

1150 TERR_CSTA_OPER_INV_CALL_ID

1151 TERR_CSTA_OPER_INV_DEV_ID

1152 TERR_CSTA_OPER_INV_CONN_ID

1161 TERR_CSTA_INCOMP_INCORR_STATE

1165 TERR_CSTA_INCOMP_NO_CALL_TO_

CLEAR

Description

Outgoing call has been barred.

Agent cannot log in at this time.

Agent cannot logout.

Agent cannot go to ready

state.

Invalid AuthCode.

Request incompatible with object.

Object is unknown.

Invalid call identifier.

Invalid device identifier.

Invalid connection identifier.

Incorrect object state.

No call to clear.

1183 TERR_CSTA_SUBRES_OUTST_LIMIT_EXC Rejects the second consecutive call party control request if it comes in less than one second after the first one.

1605 TERR_INVALIDPARTY Party in request was invalid on switch.

2100

2101

3005

TERR_SIPCS_ERR

TERR_SIPTS_PENDING_INV_TRN

TERR_UNSUCC_ROUTECALL

Unspecified kind of error.

Operation cannot be completed due to the pending INVITE transaction.

2130 TERR_SIPTS_INCOMPATIBLE_OPTION_

VALUE

Operation cannot be initiated due to current value of the configuration option.

2131 TERR_SIPTS_REFER_ENABLED_SHOULD_

BE_TRUE

Operation cannot be initiated if “referenabled=false”.

2132 TERR_SIPTS_NOT_SUPPORTED_IN_

CURRENT_SCENARIO

3002 TERR_PRIVVIOLATION

Operation is not supported in current scenario.

User doesn’t have security privilege on the switch.

TRouteCall

request was unsuccessful.

440 Framework 8.1

Chapter 6: T-Library Functionality Support Known Limitations

Known Limitations

Several known limitations result from the current SIP Server and softswitches/gateways interface:

•

Due to the specifics of gateway behavior in performing SIP

REFER

methods, support for remote agents has some limitations. In order to use remote agents, you must perform one of the two following steps:



Provision customers and remote agents to use physically separate gateways (otherwise, calls from agents to customers take shortcuts within gateways, which means that SIP Server loses track of the call and therefore cannot perform call control). Even in this configuration, direct calls between two remote agents on the same gateway are not visible to SIP Server.

Or,



Disable the SIP

REFER

method for the gateways where the remote agents are located. This enables SIP Server to see agent-to-customer and agent-to-agent calls.

•

SIP Server supports

TSetMuteOn

and

TSetMuteOff

only for established conferences, to allow for service observing.

•

SIP Server does not support User Datagram Protocol (UDP) messages of more than 16 KB in length. If SIP Server encounters a message larger than

16 KB, it truncates the message without warning. This can cause problems in scenarios that require larger UDP messages. For example, when using the Busy Lamp Field (BLF) feature, SIP Server can sometimes receive

UDP messages of up to 35 KB. In this scenario, the 16 KB UDP limitation restricts SIP Server support to a maximum of 20 monitored users over a single BLF subscription.

•

You must configure the Outbound Contact Solution to use a single-step transfer when you use

Transfer

Mode. SIP Server does not support consultation calls when it is working with the Outbound Contact Solution.

•

An

EventReleased (switch::)

message is issued when the last internal party leaves a call.

•

Third-party call control (3pcc) blind conference calls are not supported.

•

The

TDeleteFromConference

request is not supported for first-party call control (1pcc) conference calls.

•

SIP Server does not report a first-party call control (1pcc) conference with mixing on an endpoint.

•

SIP Server does not process third-party call control (3pcc) requests that have the same value for

AttributeThisDN

and

AttributeOtherDN

, even if the requests have a non-empty value for

AttributeLocation,

meaning that the destination DN (

AttributeOtherDN

) is located remotely. These requests are rejected with

EventError

with

Error Code

of

415

(

Invalid Destination DN

).

SIP Server—Deployment Guide 441

Chapter 6: T-Library Functionality Support Known Limitations

•

SIP Server does not support registering several SIP devices on the same

DN—only a single SIP device must be registered on a DN at a time.

Third-Party Equipment—Known Limitations

The known limitations when SIP Server is operating with a third-party equipment are as follows:

•

The Siemens OpenScape Voice switch is supported with the following limitations:





1pcc (first-party call control) calls are only supported starting with switch version V5.

The remote answer feature (

TAnswerCall

) is not available with version

2.0. However, the remote answer feature is supported with version 2.2.







For 3pcc calls with OpenScape Voice (HiPath 8000) switch versions,

SIP Server must be configured to use re-INVITE-based call control methods.

Genesys recommends setting the dual-dialog-enabled

configuration option to false

if Siemens optiPoint phones are used in re-INVITE mode for third-party call control (3pcc) operations.

Sometimes SIP Server cannot retrieve a call within a mixed phone environment. To avoid this problem, set the sip-hold-rfc3264

option with a proper value on the DN.

•

The following media gateways support re-INVITE

-based call transfers only:



Alcatel 7515



Cisco A5350







Cisco A5400

Asterisk

Sonus

•

When using the Paraxip gateway for Outbound IP, SIP Server cannot disable AM detection using the

TMakePredictiveCall

request. If you set the

Extension answer_type_recognition

to no_am_detection

in the

TMakePredictiveCall

request, SIP Server might still report AM as the CPD result in the

EventEstablished

that it generates for the call.

442 Framework 8.1