hints.enable_call_classification

Has the following values:

• true

• false

Call classification is received in connection.signal events.

hints.call_classification_timeout

Specifies the timeout for call classification in milliseconds. For example, 20000 = 20 seconds.

hints.disable_cdr_logging

Disables CDR Logging on Voice Portal if the value is true.

hints.disable_video

Disables video on Voice Portal if the value is true.

hints.enable_call_classification

Has the following values:

• true

• false

Call classification is received in connection.signal events.

hints.call_classification_timeout

Specifies the timeout for call classification in milliseconds. For example, 20000 = 20 seconds.

hints.call_classification_recorded_msg_timeout

Specifies the timeout for call classification recorded message in milliseconds. For example, 20000 = 20 seconds.

The timer starts after the call progress detects an answering machine greeting. That is the application receives a connection.signal event with the recorded_msg parameter. It is used to limit the amount of time that the call progress waits for the end of the answering machine message. If the timer fires, the application receives a connection.signal event with the session.connection.callprogress value set to timeout.

The default is 30 seconds.

hints.sip.from.displayname

Sets the Display Name field in the From: header of the generated INVITE.

hints.sip.to.displayname

Sets the Display Name field in the To: header of the generated INVITE.

hints.sip.call-info

Sets the Call-Info: header in the INVITE.

hints.sip.organization

Sets the Organization: header in the INVITE.

hints.sip.subject

Sets the Subject: header in the INVITE.

hints.sip.priority

Sets the Priority: header in the INVITE.

hints.sip.replaces

Sets the Replaces: header in the INVITE.

hints.sip.x-nt-gslid

Adds a URL parameter named x-nt-gslid to the sip URI in the To: header and request line.

This hint is used in propagating a GSLID on transfers and so on.

hints.sip.passertedid.displayname

Sets the Display Name field of the P-Asserted-Identity: header in the INVITE.

hints.sip.passertedid.uri

Sets the URI for the P-Asserted-Identity: header in the INVITE.

hints.sip.historyinfo

Controls the history info headers in the generated INVITE request.

        var hints = new Object();
hints.sip = new Object();
hints.sip.historyinfo = new Array();
hints.sip.historyinfo[0] = new Object();
hints.sip.historyinfo[0].user = "5551101";
hints.sip.historyinfo[0].host = "avaya.com";
hints.sip.historyinfo[0].index = "1";
hints.sip.historyinfo[1] = new Object();
hints.sip.historyinfo[1].displayname = "VDN1";
hints.sip.historyinfo[1].user = "5551101";
hints.sip.historyinfo[1].host = "avaya.com";
hints.sip.historyinfo[1].optheader = "Reason=SIP;cause=302;text=\"Moved Temporarily\"";
hints.sip.historyinfo[1].index = "1.1";
hints.sip.historyinfo[2] = new Object();
hints.sip.historyinfo[2].displayname = "Skill1";
hints.sip.historyinfo[2].user = "5552101";
hints.sip.historyinfo[2].host = "avaya.com";
hints.sip.historyinfo[2].optheader = "Reason=SIP;cause=480;text=\"Temporarily Unavailable\"";
hints.sip.historyinfo[2].index = "1.2";

hints.sip.media

Controls whether an RTP stream should be negotiated for the call.

It is used in SIP call scenarios like Best Service Routing (BSR) polling and call queueing that only require call signaling.

// Set media type empty so there is no media (rtp stream)
var hints = new Object();
hints.sip = new Object();
hints.sip.media = new Array();
hints.sip.media[0] = new Object();
hints.sip.media[0].type = "empty";

hints.sip.unknownhdr

Allows the setting of arbitrary headers in the generated INVITE.

var hints = new Object();
hints.sip = new Object();
hints.sip.unknownhdr = new Array();
hints.sip.unknownhdr[0] = new Object();
hints.sip.unknownhdr[0].name = "X-AnyHeader";
hints.sip.unknownhdr[0].value = "Value for header";

hints.call_classification_connectWhen

Has the following values:

• OnConnect

• OnProgress

hints.ConnectWhen

Has the following values:

• OnConnected

• OnProceeding

• OnAlerting

OnConnected is the default state if not specified.

For more details, see ConnectWhen.

hints.ASRRequired

Overrides the setting in VPMS. The value could be true or false.

hints.ASRLanguages

Specifies the language used in ASR. For example, en-US or en-SG.

hints.clamp_dtmf_duplex

Blocks DTMF (DTMF clamping) in either full duplex or half duplex direction.

It is used only when dtmfclamp join parameter is true.

Has the following values:

• half: Clamps DTMF from id1 to id2, and allows DTMF from id2 to id1.

• full: Clamps DTMF in both directions.

hints.MergeTimeout

Sets a time limit for the merge to complete, specified in milliseconds. If the underlying REFER w/Replaces operation doesn’t complete within the time limit, the merge will fail.

hints.DestinationURI

Sets the URI in the ReferTo: header of the REFER w/Replaces request.

hints.ConnectWhen

Contains the following values:

• OnConnected

• OnProceeding

• OnAlerting

OnProceeding is the default state if not specified.

For more details, see ConnectWhen.

hints.TransferTimeout

Indicates the time interval of finishing the transfer.

hints.AAI

Specifies the application to application info. AAI is a string containing data sent to an application on the far-end, available in the session variable session.connection.aai.

For details on the <transfer/> element, see VoiceXML elements and attributes.

hints.sip.respcode

Sets the numeric value of the SIP status code. This allows the application to override the default of 302.

hints.sip.resptext

Sets the string value of the SIP reason phrase. This allows the application to override the default of Moved Temporarily.

hints.h323

Specifies H323 call info.

hints.uui

Sets UUI info.

session.connection.protocol.sip.requestmethod

Contains the following values:

• info

• options

• update

These three values can be used with targettype "SIPEndpoint" on <send/> to determine the SIP request message.

session.connection.protocol.sip.body[xx].type

Specifies the content type for the multipart message body of the SIP request.

session.connection.protocol.sip.body[xx].msg

Specifies the content for the multipart message body of the SIP request.