PPM-in-SES

Interface Specification Document for Personal Profile Manager

03-601678Issue 1

August 2006

2006 Avaya Inc.All Rights Reserved.

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Issue 1 August 2006 3

Interface Specification for PPM. . . . . . . . . . . . . . . . . . . . . . . 5Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5PPM Server (SOAP interface) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

Operations supported . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6Data local to terminals. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24System wide parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . 25Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27Information stored by proxy on the SES server . . . . . . . . . . . . . . . 27Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29Data maintained in Communication Manager (CM) . . . . . . . . . . . . . 30Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46Endpoint Configuration JUMBO . . . . . . . . . . . . . . . . . . . . . . . 47VMON Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50

Home server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53Retry Fault . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

End Point to Personal Profile Manager Data Flows . . . . . . . . . . . . . . . . . 54On-demand information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54Profile Event Package . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

Reboot Required . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56Report Status. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57Reload Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58Detailed FNU Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59

Feature Name URI examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 59WSDL Document. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74

Contents

Contents

4 Interface Specification Document for Personal Profile Manager

Appendix A: PPM SOAP Interface for DevConnect . . . . . . . . . . . . 109Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

Proposed Solution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109PPM SOAP Service Specification. . . . . . . . . . . . . . . . . . . . . . . . . . . 111

getContactList . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112addContact . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116updateContact . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

Example: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119deleteContact . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120changePassword . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121getRegistrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122

Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123getHomeCapabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123

Error Conditions/Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127sendCommand . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127


Interface Specification for PPM

IntroductionThis document provides details of the interface between the Avaya Personal Profile Manager (PPM) Server and a client or end point, such as a SIP Phone. This document also describes the functionalities provided by the web interface, using SOAP, to the SIP Personal Information Manager (SIP PIM) appication that can be used by members of the Avaya DevConnect program. PPM is used as a repository for user-specific information that is pulled at the clients request, for example on an event such as user login. PPM is already a part of the SIP Enablement Services (SES) infrastructure. This document describes only additions in SES R3.x feature development to the existing interface. The additions for DevConnect members to the PPM Server interface as well as to the SIP PIM web interface are defined in detail in later sections of this document.

PPM Server (SOAP interface)PPM Server exposes a SOAP Remote Procedure Call (RPC) interface. (SOAP defines how an XML body of a specific format is carried in an HTTP request or response.) This interface is defined in a WSDL document and described in the following sections. A snapshot of the WSDL document can be found in the section titled WSDL Document on page 74.

The PPM server is implemented on the Apache Axis framework, which in turn runs as a service on top of Apache Tomcat software. A tool is used to build the PPM server from the actual WSDL document.

! Important:Important: Although reasonable attempts are made to align this document (including

the attached WSDL document) with the actual WSDL document, the actual WSDL document should always be used as the definitive interface specification.

Several toolkits exist that may be used to help build the client interface, such as Axis or .NET, but these toolkits are not required.

The PPM client obtains the data from PPM Server by executing the desired operations defined in the WSDL document. Note that each request to the PPM server will be authenticated using digest authentication (see RFC 2617). This means that each request may actually consist of two round-trip requests (the original request, the response with a challenge, the subsequent request containing the challenge response, and the final response). A nonce aging parameter



may be set in the PPM Server so that each request after the first request would not necessarily be challenged. If a subsequent request (e.g., for operation 2 in the following figure) is received at the PPM Server before the nonce age has expired and the credentials are otherwise correct, the request will not be challenged. The following figure shows just one example of the communication, in this case, between a Toshiba SIP Phone and the PPM Server in SES:

Operations supportedThe following sections describe the requests supported in the PPM Server, as well as the responses to those requests. In general, get requests are used to pull data from the PPM repository, while set requests are used to store data in the PPM repository.

Data local to terminals

Terminal informationThis terminal information may be used by any end point of the PPM. The terminal calls this method to store this information so that the SES administrator may view this information at a later time. This information must be propagated back to the SES master administration system.

The ContactUri parameter is used to identify which of the users contacts is storing its information. SES allows users to have multiple devices, i.e. SIP contacts, so this method must qualify which device is storing the information.

401 Unauthorized (WWW-Authenticate)

Toshiba terminal PPM Server

POST (Operation 1)

POST (Operation 1 with Authorization)

200 OK (with response)

POST (Operation 2 with Authorization)

200 OK (with response)

PPM Server (SOAP interface)


setTerminalInformation(Handle, TerminalInformation)

Request Parameter

Type Value Description

Handle String SIP URI The users primary handle

TerminalInformation TerminalInfo Container for Terminal information

TerminalInfo Type Value Description

Vendor String Maximum of 32 UTF-8 characters

Name of terminal vendor

PhoneType String Maximum of 32 UTF-8 characters

Type of terminal (e.g., model)

ProgramVersion String Maximum of 32 UTF-8 characters

Firmware/Software Version present in the terminal

MacAddress String 17 characters Hardware (MAC) address of the terminal



Example Terminal Request:

POST /axis/services/ppm HTTP/1.0Content-Type: text/xml; charset=utf-8Accept: application/soap+xml, application/dime, multipart/related, text/*User-Agent: Axis/1.1Host: 127.0.0.1Cache-Control: no-cachePragma: no-cacheSOAPAction: "setTerminalInformation"Content-Length: 1740

[email protected] [email protected] SUSHI SIP 3.0 00:D1:B6:C6:E1:7B



Example PPM Response:

Error Conditions/Faults

The following faults are sent to indicate error conditions.

Volume Settings

This operation allows a phone to get and set its Ringer Volume, Speaker Volume, and Receiver Volume settings.

getVolumeSettings(Handle)

HTTP/1.1 200 OKContent-Type: text/xml; charset=utf-8Date: Fri, 16 Jan 2004 12:40:16 GMTServer: Apache Coyote/1.0Connection: close





VolumeSet Type Value Description

RingerVolume Short {1, 2, 3, 4, 5, 6, 7, 8, 9, 10}

Relative volume on a scale of 1 10

ReceiverVolume Short {1, 2, 3, 4, 5, 6, 7, 8, 9, 10}

Relative volume on a scale of 1 - 10

SpeakerVolume Short {1, 2, 3, 4, 5, 6, 7, 8, 9, 10}

Relative volume on a scale of 1 - 10

RingerCadence Short {1, 2, 3} One of a few choices for ringer cadences

[email protected]




The following fault is sent to indicate an error condition.

setVolumeSettings(Handle, VolumeSet)


Fault Description

DatabaseError PPM cannot access the Database for reading or writing.

Request Parameter

Type Description

Handle String Users primary handle

VolumeSettings VolumeSet See VolumeSet above

[email protected] 6 3 5 3






Ringer Settings

A phone can get and set Ringer Cadence and Ringer On/Off settings with this operation.

getRingerSettings(Handle)




RingerOnOffListInfo Type Value Description

NoOfElements Integer Number of elements in the array

RingerOnOffList Array of RingerOnOffData

See RingerOnOffData below

RingerOnOffData


OnOffSet Enum {On, Off} on indicates the ringer will alert when a new call arrives; off indicates the ringer will not alert when a new call arrives

ButtonNumber Short Between 1-20 the location of the button for this ringer on/off setting

Address String SIP URI The Address of the Button

[email protected]






setRingerSettings (Handle, ListOfRingerOnOffData)





RingerOnOffData


OnOffset Enum {on, off} on indicates the ringer will alert when a new call arrives; off indicates the ringer will not alert when a new call arrives

ButtonNumber Short Between 1-20

the location of the button for this ringer on/off setting

[email protected] 1





Speed Dial List

A Speed Dial List (SDL) is accessible for each SIP phone. A phone can retrieve its Speed Dial List by invoking the getSpeedDialList request on the PPM server.

PPM always provides a mechanism for a terminal or end point to retrieve its speed dial list.

A speed dial entry has the following attributes:

Entry Number

Name (Kanji name for SUSHI)

Alias (Kana name for SUSHI)

Prefix

Address

Handle

Release: 3.0Source: SUSHIStatus: BaselinedThe speed dial list is a subset of a users contact list. Each contact in the contact list may be enabled as a speed dial entry. The user only has one speed dial list, and if the user has multiple terminals that support the getSpeedDialList method; all terminals receive the same copy of the speed dial list. This is consistent with a user having a single contact list.

When a contact is enabled as a speed dial entry, the user has the opportunity of adding a prefix in the prefix field an telephone number in the address field (as opposed to a SIP URI). It is acceptable for the prefix field to remain blank, in which case this method will return an empty tag for Prefix-TelNum.

Fault Description

Invalid Value If value to be set is not of enum type OnOffSet then this fault will be send to endpoints

Invalid Button Value If Button number value is not in the range 1-20, this fault occurs

Database Error PPM cannot access the Database for reading or writing.

Invalid Button Type The operation is not allowed on this button type.



getSpeedDialList(Handle)

For each entry in the Speed Dial List, the data shown in the following table is provided in the response:

Request Parameter

Type Description

Handle String Users handle

Response Parameter Type Description

ListOfSpeedDialListData SpeedDialListInfo See Below

SpeedDialListInfo


NoOfElements Integer SOAP Standard The number of speed dial entries

SpeedDialList Array ArrayOfSpeedDialEntry The Array of entries

SpeedDialEntry Type Value Description

Number Short 1-100 This value specifies the entry in the Speed Dial List. The Speed Dial list may contain a maximum of 100 entries, so this number can be in between 1-100.

Name String Maximum of 16 UTF-8 characters

For SUSHI, this is a 10 Kanji Character value which represents the name assigned to the button. For example, Button 1 has assigned Name Steve, Button 2 has assigned Name Sakai-San, etc. The SUSHI SIP PIM page will limit the number of characters entered.

Alias String Maximum of 16 UTF-8 characters

For SUSHI, this is 16 Kana characters to help sorting.

PrefixTelNum String Maximum of 4 digits

A prefix, such as an AAR or ARS access code, required in addition to the targets telephone number

Address String Maximum of 256 characters

A telephone number (or a general SIP URI) assigned to the button for speed dial

Handle String Maximum of 256 characters

The contacts user handle




ajbaker



One Touch Dial List

The One Touch Dial List is a set of buttons and associated addresses that will be called when the button is pressed. A phone can retrieve the One Touch Dial List by invoking the getOnetouchDialList request on the PPM server.

The one touch dial list can be managed from the SIP PIM or from the end point using the getOneTouchDialList and setOneTouchDialList methods. Note that the button location is determined from the administration of an autodial button in the station form on Communication Manager.

If an address has been provisioned for the button on Communication Manager, PPM will indicate that this button is read only [i.e. ReadOnly = true]. Read only buttons may only have their labels customized by the end user, and attempts to change the address will result in a fault. Otherwise, buttons may have their address and label customized by the end user (via setOneTouchDialList).

getOneTouchDialList(Handle)

Request Parameter

Type Description


Response Parameter Type Description

ListOfOneTouchDialData OneTouchDialListInfo See Below

OneTouchDialListInfo Type Description


OneTouchDialList Array of OneTouchButton

See OneTouchButton below

OneTouchButton Type Value Description

ButtonLocation Short 1-66 the location of button


number (or general SIP URI) assigned to the button for One Touch Dial




Label String(Optional)

Maximum of 20 UTF-8 characters

This is a label that can be displayed for the button. If no label has been set (via setOneTouchDialList), PPM will return a label formed from the Address.

ReadOnly Boolean True/False This indicates if the one touch button is read-only or not. Read-only buttons are maintained in Communication Manager and the user may not change the address of the button.

user@domain




setOneTouchDialList(Handle, ListOfOneTouchDialData)

Request Parameter Type Description

Handle String Users primary handle

ListOfOneTouchDialData OneTouchDialListInfo Container for One touch dial list array information

OneTouchDialListInfo Type Description


OneTouchDialList Array of OneTouchButton

See OneTouchButton below


ButtonLocation Short 1-66 the location of button


number (or general SIP URI) assigned to the button for One Touch Dial

Label String (Optional)

Maximum of 20 UTF-8 characters

This is a label that can be displayed for the button. If no label has been set (via setOneTouchDialList), PPM will return a label formed from the Address.







user@domain 1 11 5551212

PPM_Success

Fault Description




System wide parameters

Line Reservation Timer

A phone may obtain its Line Reservation Timer setting by invoking getLineReservationTimer. The Line Reservation Timer is used to reserve a line appearance for the designated time. This timer is set by the system administrator.

getLineReservationTimer()

Example Phone Request:


Response Parameter


ResvTimer Short 30-240 seconds Line Reservation timer value





Expired Timer Value

The getExpiredTimerVal operation allows a phone or other SIP end point to obtain the expired timer value that is set by an administrator.

getExpiredTimerVal ()

Example End Point Request:

Fault Description


Response Parameter


ExpIntVal Integer 10 864000 seconds (maximum of 10 days)

This is the value to be placed in the Expires header of the SIP REGISTER message.






Information stored by proxy on the SES server

Session History (Future)

DEFERRED FEATURE: Session history (for example, a call log) can be retrieved from the SES server via PPMS. The getCallLog request returns the session history for the last 20 incoming and last 20 outgoing calls. The response message will provide session history entries sorted according to incoming/outgoing, then date and time, with the most recent call first in the list.

getCallLog(Handle)



Note:Note: This document does not specify how data is stored in the call log, or how the call

log reflects feature interactions. Call log behavior will be described in CID 103450.


CallLogListInfo Type Description


CallLog Array of LogEntry

See LogEntry below

LogEntry Type Value Description

Direction Enum {incoming, outgoing}

Indicates whether the call was originated by the user (outgoing) or received by the user (incoming)

Status Enum {answered, unanswered}

Indicates whether the call was answered or not

CallDate Date YYYY-MM-DD Date that the session was initiated (i.e., when the INVITE was sent)

CallTime Time HH:MM:SS UTC timestamp indicating the time that the session was initiated (i.e., when the INVITE was sent)

Address String Max of 256 characters

Address of the other party (i.e., the calling party if Direction is incoming, the called party if Direction is outgoing). May be a digit string or a SIP URI.

CallDuration Time HH:MM:SS Duration of the session once Baselined (e.g., talk time).

user@domain



Data maintained in Communication Manager (CM)

A variety of configuration data is maintained in the Avaya Communication Manager. PPM acts as a front end to access this data.

Idle Line Selection

A phone or other SIP end point can obtain its idle line selection settings from PPM by invoking getIdleLineSelection.

getIdleLineSelection(Handle)

Request Parameter

Type Description


Response Parameter


LinePreferenceInfo LinePrefInfo Container of line preference information

LinePrefInfo Type Value Description

CallAppPreference Enum (LinePref)

{y, n} the Idle Appearance Preference defined for this station in CM

BridgeAppPreference Enum (LinePref)

{y, n} the Bridged Appearance Preference defined for this station in CM (As of this writing, adding a bridged appearance preference is a planned development item in CM 3.0. The name of this field for CM administration is not yet known. The possibility exists that this feature is postponed from 3.0.)







user@domain



Message Waiting Extension

A SIP phone may obtain the Message Waiting Extension setting from PPM by invoking getMessageWaitingExt.

getMessageWaitingExt(Handle)



Request Parameter

Type Description


Response Parameter


MWExt String Maximum of 7 digits the message waiting extension assigned to the station on CM (Note that the Message Waiting Extension assigned on CM is generally used to light a message waiting lamp on the telephone.)

user@domain





Auto Answer Type

A SIP phone may obtain its Auto Answer Type setting from the PPM by invoking getAutoAnswerType.

getAutoAnswerType(Handle)

Fault Description

DataNotAvailable PPM is unable to obtain the data from CM. A likely cause is that PPM was not able to obtain a response from CM before link timeout.

ConfigurationMismatch The station has not been administered on CM.

Request Parameter

Type Description


Response Parameter


AutoAnswer AutoAnsType See Below

AutoAnsType Type Value Description

AutoAnsType Enum {none, icom, acd, all}

the Auto Answer setting for this station on CM. SUSHI will only support none and icom; any other value will be treated like none.







Dial Plan

A phone may obtain a simplified Dial Plan with this operation. The phone could use a dial plan to determine the end of dialing (i.e., when to send the INVITE) and application of a second dial tone (e.g., such as when using ARS, like dialing 9 to get an outside line). The Dial Plan will be represented as a list of terms, where each term is a nearly generic regular expression form.

user@domain



The rules for each term are:

x = any digit

Z = play dialtone

+ = 1 or more instances of the preceding character, implied inter-digit timer

Some sample terms for a dialplan:

538xxxx = a 7-digit number starting with 538

9Z1xxxxxxxxxx = ARS access to a long distance number in North America (dialtone after the 9)

9Z011xxxx+ = ARS access to an international number of unknown length, but at least 7 digits (including 011)

getDialPlan(Handle)

Request Parameter

Type Description


Response Parameter


DialPlanData DialPlanListInfo Container for all dial plan information

DialPlanListInfo Type Value Description

DialPlanDomain String 36 characters The domain of the dial plan. Currently this is same as the enterprise SIP domain.


DialPlan Array of Strings

Expression representing rules described above

Each string i n the array represents a single term of the CM dial plan



Example SUSHI Request:




user@domain



Button Assignment

A SIP phone or device may obtain its button assignments by invoking getButtonAssignments.

getButtonAssignments(Handle)

Request Parameter

Type Description


Response Parameter Type Value Description

ListOfButtonAssignments ButtonListInfo Container for Button Assignments information

ButtonListInfo Type Description

NoOfElements Integer Number of elements in array

ButtonAssignment Array of ButtonData

See ButtonData below

ButtonData Type Value Description

Location Short 1 66 This value specifies the number (location) of button. SUSHI phones will be a CM set type that provides 20 buttons. Other CM set types could support as many as 66 buttons.

ButtonType Enum {call-appr, brdg-appr, aut-msg-wt, FNU, autodial}

This specifies which type of Button

Label String Maximum of 20 UTF-8 characters

A string that can be displayed for the button. If no label has been defined, PPM will provide a default label from known data (for example, from the extension for a bridged appearance).

LineID Short SOAP Standard This gives the LineID value, or no value if a line ID is not relevant for the button type. For a bridged appearance button, this is the Button subfield found on CM.



Address String Max 256 characters If the button type is brdg-appr, the bridge extension goes here. If the button type is autodial, the destination digits/URI goes here. If auto-msg-wt, the destination extension goes here. For FNU, see FNUInfo below. For some FNUs (e.g. call-fwd), the source extension, if different than the users, is specified here.

FNUType String A pre-defined string unique to each supported feature

See FNU Description below

App Boolean If true, the endpoint must request a line appearance (+line) on the Contact header in an INVITE with this FNU.

Media Boolean If true, the endpoint must include an SDP offer in an INVITE with this FNU.

FNUInfo Array of FNUData

Varies by FNU See FNU Description below

FNUData Type Description

Parameter String A parameter to the FNU (as presented in FNUType). If the parameter requires completion by the client, it is the parameter string ending with = (e.g., avaya-cm-action=). If the parameter is complete (e.g., avaya-cm-group=3), the client would use the parameter as is.

Required Boolean If true, the client must include this parameter. If false, the client may include the parameter if it is able to complete the parameter or provide the feature processing.

MustComplete Enum If none, the parameter is already complete. If offon, the endpoint must complete the parameter with either off or on. If address, the endpoint must complete the parameter with an address (e.g., telephone number or extension). If number, the endpoint must complete the parameter with a number. If dialcode, the endpoint must complete the parameter with a dial code.

ButtonData Type Value Description



FNU DescriptionThe following table shows the list of features accessible via FNU in Communication Manager. Many of the FNUs require additional parameters. A yes value in the column labeled MustComplete indicates that the endpoint must complete the indicated parameter. A no value in the column labeled MustComplete indicates that the FNU parameter provided by PPM in FNUInfo is fully qualified and requires no further addition from the endpoint.

Feature Name FNUType App Media FNUInfo

Parameter Required MustComplete

Automatic call back

avaya-cm-fnu=auto-callback false false avaya-cm-action={on|off} true offon

Automatic intercom

avaya-cm-fnu=auto-intercom true true avaya-cm-group={group}

avaya-cm-dial-code={code}

true

true

none

none

Busy Indicator avaya-cm-fnu=busy-indicator true true avaya-cm-destionation={address} true address

Call forwarding allSee note 1See note 4

avaya-cm-fnu=call-forwarding-all

true true avaya-cm-destination={address}

avaya-cm-action={on|off}

false

true

address

offon

Call forwarding busy/no answerSee note 1See note 4

avaya-cm-fnu=call-forwarding-busy-no-answer

true true avaya-cm-destination={address}

avaya-cm-action={on|off}

false

true

address

offon

Call parkSee note 3

avaya-cm-fnu=call-park false false

Call unpark (answer back)See note 2See note 3

avaya-cm-fnu=call-unpark true true avaya-cm-extension={address} false address

Call pick-up Group

avaya-cm-fnu=call-pickup true true

Call pickup Directed

avaya-cm-fnu=call-pickup-directed

true true avaya-cm-extension={address} false address

Call pickup Extended group See note 2

avaya-cm-fnu=call-pickup-extended

true true avaya-cm-pickup-number={num} false number

Per call calling party number blockSee note 3

avaya-cm-fnu=calling-party-block

true true avaya-cm-destination={address} false address

Per call calling party number unblockSee note 3

avaya-cm-fnu=calling-party-unblock

true true avaya-cm-destination={address} false address

Dial Intercom avaya-cm-fnu=dial-intercom true true avaya-cm-group={group}

avaya-cm-dial-code={code}

true

false

none

dialcode

1 of 2



Note 1: When deactivating feature (avaya-cm-action=off), other parameters are not included.

Note 2: No button is defined on CM for this feature, so DevConnect members will never see this from PPM. The FNU is supported by Avaya, and therefore it is listed here for completeness.

Note 3: Unused, but supported by Avaya so included here for completeness.

Note 4: Session Description Protocol (SDP) and the +line parameter in the Contact header must be included if the optional FNU parameter is not present in the INVITE. Conversely, if the optional FNU parameter is included in the INVITE, then the INVITE should not include SDP or the +line parameter.

Exclusion See note 3

avaya-cm-fnu=exclusion false false avaya-cm-action={on|off} true offon

Off-PBX call See note 3

avaya-cm-fnu=off-pbx false false avaya-cm-action={on|off} true offon

Off hookSee note 2

avaya-cm-fnu=off-hook true false

Last number dialedSee note 3

avaya-cm-fnu=last-number-dialed

true true

Malicious Call Trace See note 3

avaya-cm-fnu=mct false false

Malicious Call Trace CancelSee note 2See note 3

avaya-cm-fnu=mct-cancel false false

One Touch Recording See note 1See note 3

avaya-cm-fnu=one-touch-recording

false false avaya-cm-action={on|off}

avaya-cm-extension={address}

true

true

offon

address

Priority callSee note 3

avaya-cm-fnu=priority-call true true avaya-cm-destination={address} false address

Send all calls avaya-cm-fnu=sac false false avaya-cm-action={on|off} true offon

Transfer to voice mailSee note 2

avaya-cm-fnu=transfer-to-voicemail

false false

Whisper page avaya-cm-fnu=whisper-page true true avaya-cm-extension={address} false address

Feature Name FNUType App Media FNUInfo

Parameter Required MustComplete

2 of 2



Examples:Automatic callback For an automatic callback button in location 9, PPM returns the following structure:

When the user presses the automatic callback button to activate the automatic callback feature, the endpoint would send an INVITE with the request URI of:

sip:handle@domain;avaya-cm-fnu=auto-callback;avaya-cm-action=on

It is the endpoints responsibility to complete the parameter avaya-cm-action=on because the MustComplete value for the avaya-cm-action parameter is offon.

Automatic Intercom For an automatic intercom button in location 8, PPM returns the following structure:

When the user presses the automatic callback button to activate the automatic callback feature, the endpoint would send an INVITE with the request URI of:

sip:handle@domain;avaya-cm-fnu=auto-intercom;avaya-cm-group=3;avaya-cm-dial-code=2

In this case, PPM provides all values for the FNU parameters, so the endpoint need not complete the FNU parameters (the MustComplete value for both parameters is none). When a CM administrator adds an auto-icom button to a station, the administrator must also enter a group and dial code. When PPM gets the button information from CM, it also gets the group and dial code.

Location 9

ButtonType FNU

LineID Null

Address Null

FNUType avaya-cm-fnu=auto-callback

FNUInfo {avaya-cm-action=, true, offon}

Location 8

ButtonType FNU

LineID Null

Address Null

FNUType avaya-cm-fnu=auto-intercom

FNUInfo {avaya-cm-group=3, true, none}{avaya-cm-dial-group=2, true, none}




user@domain





Soft Menu Keys Assignment

A SIP phone or other device may obtain its soft menu key assignments with the getSoftMenuKeysAssignments operation.

getSoftMenuKeysAssignments(Handle)

See Dial Plan and Button Assignment on page 37 for definition of Button Data.

Fault Description



Request Parameter

Type Description


Response Parameter


SoftMenuKeyList ButtonListInfo Container for SoftMenuKeys information

ButtonListInfo Type Description


ButtonList Array of ButtonData

See ButtonData above



Example SUSHI Request:


user@domain





Report Status

Due to the stateless nature of web environments, both in the personal profile manager and the SES web interface, it is necessary to temporarily store information received so that the administrator can view it. To meet the maintainability and serviceability guidelines, it is necessary for a technician to get the status of a SIP end point. To achieve this, a get status link is clicked from the Registered Devices page present at each home node in the network. Clicking this link causes the web interface to trigger a notify to that end point stating that its status is needed. The end point in turn calls the Report Status method in PPM.

The PPM takes the input from the Report Status method and stores the information in a temporary location. This operation may take a second to be completed. To simulate a real time experience in the web interface, the View Status page should examine the database every 2 seconds until a record is added for the device in question.

reportStatus(Handle, OptimExtension, IPAddress, HostName, SwitchHookStatus, PresenceStatus)

Fault Description



Request Parameter


Handle String SIP URI The users primary handle

OptimExtension String This terminals extension

IPAddress String This terminals IP address

HostName String This terminals host name

SwitchHookStatus Enum {On, Off} Indicates if the terminal is currently on-hook or off-hook

PresenceStatus String This terminals current presence status



Example client request:


Endpoint Configuration JUMBO

A method exists that can be used to obtain endpoint configuration in one request. The getEndpointConfiguration operation combines the getIdleLineSelection, getMessageWaitingExt, getAutoAnswerType, getLineReservationTimer, getExpiredTimerVal, getRingerSettings, getVolumeSettings, getButtonAssignments, getOneTouchDialList, and getSoftMenuKeyAssignments operations.

getEndpointConfiguration(Handle)

user@domain 1001 135.8.68.110 ipphone.usae.avaya.com ajbaker on-hook available



Response Parameter


ConfigInfo EndpointConfigInfo Container for Endpoint configuration information

EndpointConfigInfo Type Value Description

VolumeSettings VolumeSet See VolumeSet. This terminal information may be used by any end point of the PPM. The terminal calls this method to store this information so that the SES administrator may view this information at a later time. This information must be propagated back to the SES master administration system.)

ListOfRingerOnOffData RingerOnOffListInfo See RingerOnOffData in getVolumeSettings(Handle) )

ResvTimer short 30 - 240 seconds Line Reservation timer value

ExpIntVal Integer 10 864000 seconds (maximum of 10 days)

This is the value to be placed in the Expires header of the SIP REGISTER message.

LinePreferenceInfo LinePrefInfo See Idle Line Selection

MWExt String Maximum of 7 digits

the message waiting extension assigned to the station on CM (Note that the Message Waiting Extension assigned on CM is generally used to light a message waiting lamp on the telephone.)

AutoAnswer AutoAnsType {none, icom, acd, all}

the Auto Answer setting for this station on CM

ListOfOneTouchDialData OneTouchDialListInfo See OneTouchButton -- A Speed Dial List is accessible for each terminal. A SIP phone or device can retrieve its Speed Dial List by invoking the getSpeedDialList request on the PPM server.

ListOfButtonAssignments ButtonListInfo See ButtonData (section getDialPlan(Handle))

SoftMenuKeyList ButtonListInfo See ButtonData (getDialPlan(Handle))

DialPlanData DialPlanListInfo See Dial Plan (section Dial Plan)

ListOfSpeedDialData SpeedDialListInfo See SpeedDialEntry (section getRingerSettings(Handle))

CallLog CallLogInfo See LogEntry (section Session History (Future)). Note this was deferred.

VmonInformation VmonInformation See VMON Information (2.1.6)



VMON Information

The getVMONInformation() method may be used as a stand alone request, or the end point may receive the same information from the JUMBO (getEndPointConfiguration()). This request provides the end point with the RTCP Monitor IP Address, Port Number, and Report Period (in seconds).

The VMON information is administered in the SES administration system in the 3.0 release of SES. In subsequent releases, the VMON information will be retrieved from the network region tables in Communication Manager.

Note that the same VmonInformation SOAP Struct data type is used in the getEndPointConfiguration response.

getVMONInformation()


VmonInformation Type Value Description

RtcpServer String IP address The IP address of the RTCP Monitor

VmonPort Integer Port The port of the RTCP Monitor

ReportPeriod Integer Seconds How frequently the end point should be sending the receiver report






Home serverThe Avaya PPM team recommends that end points call the getHomeServer method only once during the initial installation and persist the location until one of the following circumstances:

The end point receives a fault from any other PPM method that states the user is not known. The end point should then call the getHomeServer method to verify the users location. This occurs when the digest is still valid but the user was just moved.

The end point receives a notification stating it needs to update its location.

The end point fails authentication with the PPM or the SIP proxy. For some reason if the user was moved from one SES to another and the end point isnt notified, the end point may wish to re-validate its home server as an automated run-time troubleshooting step.

The end point looses the location information (e.g. non data preserving firmware upgrade)



This method is used when the phone has no record of its location such as during installation time. If a customer deploys multiple SES servers on the same subnet, the existing DHCP mechanism for managing service location does not provide the granularity necessary for two clients on the same subnet to use two different servers in a straight forward and automated manner.

The getHomeServer method is used to retrieve the location of the service that the user is supposed to be using. This method may be executed against any SES home in the network and will return the exact same information regardless of which home is local to the request. Once the service location is retrieved, the end point may proceed with SIP registration and initialization of data with its home PPM.

getHomeServer(handle)

Request Type Value Description

Handle String URI The handle is the primary handle of the user, maybe in the form of user or user@domain

Response Type Value Description

ServerInfo HomeServerInfo Container of HomeServer Information

HomeServerInfo Type Value Description

ppmServer String HTTP URL The URL for the users home PPM server.

sipServer String Host The hostname or IP address of the users home proxy

sipPort Integer Usually 5060 The port that the sip server is listening on

sipDomain String The SIP domain as provisioned in SES admin

The end point should always send the register for user@domain, where domain is the SIP domain and not the hostname of the SIP server.

transportList ArrayOfTransportdata Array of transport information





Transportdata Type Value Description

transportName String {tls, tcp, udp} Name of the transport

transportPort Integer {5060, 5061} The port of the transport. In this release, it will always be either 5060 or 5061

user@domain





If the authentication fails for the getHomeServer method, this implies that either the password is invalid or that the user name is invalid.

If the user and password are correct, the user will be authenticated and any errors in retrieving the server information will be returned in the ServiceError SOAP Fault.

Retry Fault

Due to the complex relationships the PPM has with both the Converged Communications Server and Communication Manager systems, there may be instances where the PPM must throttle requests to service many simultaneous requests. In these cases, the PPM may return a SOAP stating a number of seconds that the end point should re-try the exact same request. This feature allows PPM to continue servicing end points that it may not have been able to before.

Example SOAP Fault

Notice in this SOAP fault that the PPM returns the standard SOAP fault structure. The faultstring is Retry and the detail contains an integer. The client should retry the request after waiting that integers time in seconds.

Fault Description

ServiceError If there is an error in retrieving the location from the back end system

soap:Server Retry

30



End Point to Personal Profile Manager Data Flows

On-demand informationThe Personal Profile Manager does not restrict how or when the terminal is to fetch its data. The terminal may choose to retrieve its configuration when it initially powers on and [SIP] registers with SES, and every subsequent [SIP] re-register, or every other re-register, or not at all. The only time a terminal must retrieve information from PPM is on a user event, e.g. the user clicks View Call Log on their terminal, or from a profile event package event. This section specifies the data flow for when a user interacts with their terminal and the terminal must retrieve information from PPM.

In all instances, the terminal must include credentials when it retrieves information from PPM on demand. When the user presses the button to View Call Log, the terminal sends a SOAP request to PPM, including the HTTP authentication credentials, and executes the getCallLog SOAP method. The PPM forms the SOAP response and sends it back to the terminal. The terminal must process the SOAP body and format the information for the display on the terminal.

Feature Requests Data from PPM

ToshibaPhone PPM

getCallLog

getCallLogResponse

User clicksView Call Log

Call Logdisplayedon terminal

End User



Profile Event PackageThe Profile Event Package is an Avaya proprietary event package to notify SIP end points of profile events. There are three profile events:

Reboot Required (as demanded by system administrator via SES maintenance interface)

Report Status (as demanded by system administrator via SES maintenance interface)

Reload Profile (when users profile changes on SES, or system administrator demands via SES maintenance interface)

The phone must subscribe to the profile event package as one of its initializing steps. By subscribing to this event package, the phone will gain several desirable features. First, the administration may use the SES administration system to reboot one or more end points. The administration system triggers the NOTIFY to the terminal, and the phone or device reboots.

The administrator may also wish to have an on-demand status of the end point. In this case, the administrator clicks on Get Status for a terminal in the SES administration system. A NOTIFY is generated to that specific end point, and the end point calls the report Status method in the Personal Profile Manager. The administrator may then view what the terminal reported back as its status.

The final feature is for user experience. If some information in the users profile changes, e.g. the user changes information from the SIP Personal Information Manager or the administrator changes some information, the end users terminal must be updated to reflect this information. The terminal will receive a notify whenever some information in the profile has changed. The terminal should then execute all of its initialization methods i.e. getEndPointConfiguration, getDialPlan, and getSpeedDialList. The Notify includes a timestamp of the most recent profile. The endpoint can check this timestamp, if its profile is up to date then it need not execute the initialization methods.

All of these events are handled under one subscription to the Profile Event Package. The terminal should subscribe as part of its initialization process. Note that the notifications are SIP NOTIFY messages with XML bodies.



Just one example of a SUBSCRIBE message might be:

Reboot Required

Whenever the following notify is received, the terminal should reboot at its earliest convenience (e.g. when the user is no longer on an active call).

SUBSCRIBE sip:[email protected] SIP/2.0From: ;tag=random1To: Call-Id: [email protected]: 1 SUBSCRIBEContact: Event: avaya-ccs-profileAccept: application/profile+xmlExpires: 3600Max-Forwards: 70User-Agent: Toshiba/2.1.8 (VxWorks)Via: SIP/2.0/UDP 10.0.75.2branch=z9hG4bK=random-2Route: Content-Length: 0

NOTIFY sip:[email protected] SIP/2.0Call-ID: [email protected]: 2 NOTIFYFrom: ;tag=random2To: ;tag=random1Via: SIP/2.0/UDP 10.0.0.100;branch-id=z9hG4bK-random-ccs1SIP/2.0/TLS 10.0.0.200;branch-id=z9hG4bK-random-cm1Content-Length: 22Content-Type: application/profile+xmlContact: Max-Forwards: 69User-Agent: Avaya Communication Manager v1.0Event: avaya-ccs-profileSubscription-State: active;expires=3600Record-Route:

Reboot [timestamp]



Report Status

Whenever the following notify is received, the terminal should call the reportStatus method in the Personal Profile Manager.

Reload Profile

The profile code assists the end point in making a decision on which PPM methods to call when it receives the reload profile notify.


Report [timestamp]

Profile Code (Enum)

Description

0 The location has changed, the end point should call getHomeServer and re-initialize fully

1 Button data changed, the phone should execute getEndPointConfiguration

2 Contact list changed. The phone should execute getcontactList and getSpeedDialList.

3 Away Messages changed

4 Permissions changed.



The phone is only suggested to call the methods that correspond to the profile code. If the phone does not support a speed dial list or a contact list, then a NOTIFY with an event code of 2 should be ignored. Similarly, the phone may choose to re-initialize all data regardless of the profile code.

The lack of a profile code indicates that the phone should re-initialize fully. Note in the first release, the SUSHI end point is expected to only receive 0 or 1 for a profile code.

Figure 1: Reload profile Notify

References1. Gateway Control Protocol Version 1 (aka H.248.1), RFC 3525, http://www.ietf.org/rfc/

rfc3525.txt

2. HTTP Authentication: Basic and Digest Access Authentication, RFC 2617, http://www.ietf.org/rfc/rfc2617.txt

3. Simple Object Access Protocol (SOAP) 1.1, http://www.w3.org/TR/SOAP/

4. Web Services Definition Language (WSDL) 1.1, http://www.w3.org/TR/wsdl

5. Apache Web Services Project, Axis, http://ws.apache.org/axis/


Reload [timestamp] 0

Detailed FNU Description



Feature Name URI examplesIn all of these examples:

1111 is the CM extension associated with the endpoint that is sending the INVITE FNU.

2222 is used where the feature is activated on another CM extension. The actual value of 2222 is provided to the endpoint by PPM, and is placed by the endpoint in the user part of the Request-URI. If there is no example here using 2222 as the user part of the Request-URI for a particular FNU, then that FNU only accepts the endpoints own extension in that field (i.e., 1111).

3333 is a any other CM extension (e.g., the object of a directed call pickup).

4444444 is any number within the CM dial plan, which may be either on or off switch.

Parameters are marked as either optional (O) or mandatory (M). This distinction is relative to the inclusion of these parameters in the INVITE. Some of these parameters might still be mandatory to CM. If an optional parameter is not sent by the endpoint in the INVITE, but is required by CM, CM will provide dial tone to the user and collect the required digits.

Parameters may be supplied in any order. The order in this document is recommended.

Whenever available, the value of a parameter is extracted from a CM button by PPM, and then provided by PPM to the endpoint (e.g., avaya-cm-group). For values not available from CM, and therefore not provided by PPM to the endpoint, the endpoint will need to do one of the following:

determine the value algorithmically (e.g., avaya-cm-action),

collect the value from the user (e.g., avaya-cm-destination),

or, if an optional parameter, leave it off and let CM collect it.

The endpoint must send SDP in the INVITE for those FNUs where indicated below. Whenever sending SDP, the endpoint must also request a line appearance (+line) on the Contact header.

Automatic Call Back

INVITE sip:[email protected];avaya-cm-fnu=auto-callback;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=auto-callback;avaya-cm-action=off SIP/2.0



Parameters:

Automatic Intercom

INVITE sip:[email protected];avaya-cm-fnu=auto-intercom;avaya-cm-group=9;avaya-cm-dial-code=12 SIP/2.0

Parameters:

Name Values M|O PPM

avaya-cm-action on or off M No

Authorization: This request must be authorized by the class of service of the extension.

Description: Automatic Call Back activates or deactivates the Automatic Call Back feature for an endpoints own extension. When activated, this feature allows a user who placed a call to a busy or unanswered telephone to be called back automatically when the called telephone becomes available to receive a call.

CM button: auto-cback

Feature package: Yes

SDP required: No

Name Values M|O PPM

avaya-cm-group Intercom group number from 1 to 32 M Yes

avaya-cm-dial-code One or two digit dial code assigned to the extension to be called

M Yes



Call Forwarding All Calls, of the endpoints own (1111) extension

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-all;avaya-cm-destination=4444444;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-all;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-all;avaya-cm-action=off SIP/2.0

Parameters:

Authorization: An endpoint can use this FNU only if the endpoint has a auto-icom button administered on CM with matching group number and dial code.

Description: Automatic Intercom places a call to the station associated with the button. The called user receives a unique alerting indication. The endpoint extension and destination extension must be in the same intercom group. This feature is exactly like Dial Intercom except for the way that the dial code is specified. PPM can provide the dial code for Automatic Intercom, but not for Dial Intercom.

CM button: auto-icom Gr: 9 DC: 12

Feature package: No

SDP required: Yes

Name Values M|O PPM


avaya-cm-destination Any number within the CM dial plan, to which this endpoint is being forwarded.

O No

Authorization: This example shows the use of this FNU on the endpoints own extension. It must be authorized by the extensions class of service. See the next case for how to apply this feature to another extension.

Description: Activates or deactivates Call Forwarding All Calls.



Call Forwarding All Calls, of another endpoints (2222) extension

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-all;avaya-cm-destination=4444444;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-all;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-all;avaya-cm-action=off SIP/2.0

Parameters:

Call Forwarding Busy/Dont Answer, of the endpoints own (1111) extension

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-busy-no-answer;avaya-cm-destination=4444444;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-busy-no-answer;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-busy-no-answer;avaya-cm-action=off SIP/2.0

CM button: call-fwd Ext: (left blank)


SDP required: Only if avaya-cm-action=on and avaya-cm-destination not specified.

Name Values M|O PPM



O No

Authorization: An endpoint can use this FNU on another extension only if the endpoint has a call-fwd Ext: 2222 button administered on CM.

Description: Activates or deactivates Call Forwarding All Calls, on the extension specified in the user part of the Request-URI.

CM button: call-fwd Ext: 2222





Parameters:

Call Forwarding Busy/Dont Answer Calls, of another endpoints (2222) extension

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-busy-no-answer;avaya-cm-destination=4444444;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-busy-no-answer;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-forwarding-busy-no-answer;avaya-cm-action=off SIP/2.0

Parameters:

Name Values M|O PPM



O No

Authorization: This example shows the use of this FNU on the endpoints own extension. It must be authorized by the extensions class of service.See the next case for how to apply this feature to another extension.

Description: Call Forward Busy/Dont Answer activates and deactivates call forwarding for calls when the extension is busy or the user does not answer.

CM button: cfwd-bsyda Ext: (left blank)



Name Values M|O PPM



O No



Call Park

INVITE sip:[email protected];avaya-cm-fnu=call-park SIP/2.0

Call Unpark (Answer-Back)

INVITE sip:[email protected];avaya-cm-fnu=call-unpark;avaya-cm-extension=3333 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-unpark SIP/2.0Parameters:

Authorization: An endpoint can use this FNU on another extension only if the endpoint has a cfwd-bsyda Ext: 2222 button administered on CM

Description: Call Forward Busy/Dont Answer activates and deactivates call forwarding for calls when the extension is busy or the user does not answer, on the extension specified in the user part of the Request-URI.

CM button: cfwd-bsyda Ext: 2222



Parameters: None.

Authorization: None.

Description: Allows the user to place the current call in the call park state so it can be retrieved from another phone.

CM button: call-park

Feature package: No

SDP required: No

Name Values M|O PPM

avaya-cm-extension The CM extension where the call is parked.

O No



Call Pickup (group)

INVITE sip:[email protected];avaya-cm-fnu=call-pickup SIP/2.0

Directed Call Pickup

INVITE sip:[email protected];avaya-cm-fnu=call-pickup-directed;avaya-cm-extension=3333 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-pickup-directed SIP/2.0Parameters:


Description: Allows the user to retrieve a call parked at another extension. The endpoint will be connected to the parked call.

CM button: None. Accessed on the CM only via a FAC.

Feature package: No

SDP required: Yes

Parameters: None.

Authorization: The endpoint must be a member of a pickup group and can only pickup calls to other group members.

Description: Allows the user to answer a call that is ringing in the users pickup group.

CM button: call-pkup

Feature package: No

SDP required: Yes

Name Values M|O PPM

avaya-cm-extension The CM extension where the call is alerting.

O No



Extended Group Call Pickup

INVITE sip:[email protected];avaya-cm-fnu=call-pickup-extended;avaya-cm-pickup-number=3 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=call-pickup-extended SIP/2.0Parameters:

Calling Party Number Block

INVITE sip:[email protected];avaya-cm-fnu=calling-party-block;avaya-cm-destination=4444444 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=calling-party-block SIP/2.0

Authorization: The endpoints need not be members of a group, but directed call pickup must be authorized by the class of restriction for both endpoints.

Description: Directed call pickup allows the user to answer a call ringing at another extension without having to be a member of a pickup group.

CM button: dir-pkup

Feature package: No

SDP required: Yes

Name Values M|O PPM

avaya-cm-pickup-number The pickup number from 1 to 24. O No

Authorization: The endpoint must be a member of a pickup group, and that pickup group must be a member of an extended pickup group, which must also include the group of the endpoint whose phone is being picked up.

Description: Enables call pickup groups to answer calls directed to another call pickup group.


Feature package: No

SDP required: Yes



Parameters:

Calling Party Number Unblock

INVITE sip:[email protected];avaya-cm-fnu=calling-party-unblock;avaya-cm-destination=4444444 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=calling-party-unblock SIP/2.0Parameters:

Dial Intercom

INVITE sip:[email protected];avaya-cm-fnu=dial-intercom;avaya-cm-group=9;avaya-cm-dial-code=12 SIP/2.0

Name Values M|O PPM

avaya-cm-destination Any number within the CM dial plan, to which this call is being directed.

O No


Description: Blocks the sending of the calling party number for one call.

CM button: cpn-blk

Feature package: No

SDP required: Yes

Name Values M|O PPM


O No


Description: Deactivates calling party number (CPN) blocking and allows the CPN to be sent for a single call.

CM button: cpn-unblk

Feature package: No

SDP required: Yes



INVITE sip:[email protected];avaya-cm-fnu=dial-intercom;avaya-cm-group=9 SIP/2.0

Parameters:

Drop

INVITE sip:[email protected];avaya-cm-fnu=drop SIP/2.0

Exclusion

INVITE sip:[email protected];avaya-cm-fnu=exclusion;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=exclusion;avaya-cm-action=off SIP/2.0

Name Values M|O PPM

avaya-cm-group Intercom group number from 1 to 32 M Yes

avaya-cm-dial-code One or two digit number O No

Authorization: An endpoint can use this FNU for a intercom group that matches an administered CM button for this extension (as shown below).

Description: Dial Intercom places a call to the station associated with the button. The called user receives a unique alerting indication. The endpoint extension and destination extension must be in the same intercom group. This feature is exactly like Automatic Intercom except for the way that the dial code is specified. PPM can provide the dial code for Automatic Intercom, but not for Dial Intercom.

CM button: dial-icom Grp: 9

Feature package: No

SDP required: Yes

Parameters: None.


Description: Allows users to drop calls. Users can drop calls from automatic hold or drop the last party they added to a conference call.

CM button: drop

Feature package: No

SDP required: No



Parameters:

Off-PBX Call

INVITE sip:[email protected];avaya-cm-fnu=off-pbx;avaya-cm-action=on SIP/2.0INVITE sip:[email protected];avaya-cm-fnu=off-pbx;avaya-cm-action=off SIP/2.0Parameters:

Name Values M|O PPM


Authorization: This request always applies to the endpoints own extension. Automatic exclusion must be authorized by the extensions class of service.

Description: Exclusion allows multiappearance telephone users to keep other users with appearances of the same extension from bridging onto an existing call. If the user activates exclusion button while other users are already bridged onto the call, the other users are dropped. There are two means of activating exclusion.

Manual Exclusion when the user presses the exclusion button (either before dialing or during the call).

Automatic Exclusion as soon during a call, the user presses the exclusion button.

CM button: exclusion

Feature package: No

SDP required: No

Name Values M|O PPM


Authorization: This request always applies to the endpoints own extension.

Description: This FNU provides the capability to enable and disable the extension of an EC500 call. The EC500 station must have one or more administered bridged appearances on the SIP phone for this FNU to have the desired effect.

CM button: ec500


SDP required: No



Off Hook

INVITE sip:[email protected];avaya-cm-fnu=off-hook SIP/2.0

Last Number Dialed

INVITE sip:[email protected];avaya-cm-fnu=last-number-dialed SIP/2.0

Malicious Call Trace

INVITE sip:[email protected];avaya-cm-fnu=mct SIP/2.0INVITE sip:[email protected];avaya-cm-fnu=mct-cancel SIP/2.0

Parameters: None.


Description: Reserves a line appearance.

CM button: None.

Feature package: No

SDP required: No, but +line is required

Parameters: None.


Description: Last Number Dialed (redial) originates a call to the number last dialed by the station.

CM button: last-numb

Feature package: No

SDP required: Yes

Parameters: None.

Authorization: Must be authorized by the endpoints class of restriction.

Description: Malicious Call Trace Activation: sends a message to the MCT control extensions that the user wants to trace a malicious call. MCT activation also starts recording the call, if your system has a MCT voice recorder.

CM button: mct-act (to activate). Only a FAC to cancel.

Feature package: No

SDP required: No



AUDIX One-Step Recording

INVITE sip:[email protected];avaya-cm-fnu=one-touch-recording;avaya-cm-extension=3333;avaya-cm-action=on SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu= one-touch-recording;avaya-cm-action=off SIP/2.0

Parameters:

Priority Call

INVITE sip:[email protected];avaya-cm-fnu=priority-call;avaya-cm-destination=4444444 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=priority-call SIP/2.0Parameters:

Name Values M|O PPM


avaya-cm-extension The CM extension of an AUDIX hunt group

M Yes

Authorization: An endpoint can use this FNU on another extension only if the endpoint has a CM button audix-rec button with a matching extension.

Description: This features allows a station user to start and end the recording of an in-progress conversation using the AUDIX system recording facility. Note: avaya-cm-extension is optional when avaya-cm-action is off (because a station can only have one of these buttons).

CM button: audix-rec Ext: 3333

Feature package: No

SDP required: No

Name Values M|O PPM


O No



Send All Calls, of the endpoints own (1111) extension

INVITE sip:[email protected];avaya-cm-fnu=sac;avaya-cm-action=on SIP/2.0INVITE sip:[email protected];avaya-cm-fnu=sac;avaya-cm-action=off SIP/2.0Parameters:

Send All Calls, of another endpoints (2222) extension

INVITE sip:[email protected];avaya-cm-fnu=sac;avaya-cm-action=on SIP/2.0INVITE sip:[email protected];avaya-cm-fnu=sac;avaya-cm-action=off SIP/2.0Parameters:


Description: Priority Calling allows a user to place priority calls or change an existing call to a priority call.

CM button: priority

Feature package: No

SDP required: Yes

Name Values M|O PPM


Authorization: This example shows the use of this FNU on the endpoints own extension. No authorization is required. See the next case for how to apply this feature to another extension.

Description: Send All Calls allows users to temporarily direct all incoming calls to coverage regardless of the assigned call-coverage redirection criteria.

CM button: send-calls Ext: (left blank)


SDP required: No

Name Values M|O PPM




Transfer to Voice Mail

INVITE sip:[email protected];avaya-cm-fnu=transfer-to-voicemail SIP/2.0

Whisper Page Activation

INVITE sip:[email protected];avaya-cm-fnu=whisper-page;avaya-cm-extension=3333 SIP/2.0

INVITE sip:[email protected];avaya-cm-fnu=whisper-page SIP/2.0Parameters:

Authorization: An endpoint can use this FNU on another extension only if the endpoint has a send-calls Ext: 2222 button administered on CM.

Description: Send All Calls allows users to temporarily direct all incoming calls to coverage regardless of the assigned call-coverage redirection criteria. Applied to another extension.

CM button: send-calls Ext: 2222


SDP required: No

Parameters: None.


Description: Allows coverage to transfer the caller to the original call recipients voice mail where the caller can leave a message.


Feature package: No

SDP required: No

Name Values M|O PPM

avaya-cm-extension The CM extension to which you want to whisper

O No



WSDL DocumentThe following WSDL document (formatted for readability) defines the interface with the PPM server for the data described above. This WSDL document is included here for convenience; note that the final, normative WSDL document would be accessible from an Avaya web site or through the software interface.

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Authorization: The user must have a class of restriction (COR) that allows intraswitch calling to use whisper paging, and the extension to whom you are whispering must not have blocked whispers.

Description: Whisper Page Activation allows a user to make and receive whisper pages. A whisper page is an announcement sent to another extension who is active on a call where only the person on the extension hears the announcement; any other parties on the call cannot hear the announcement.

CM button: whisp-act

Feature package: No

SDP required: Yes

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90 Interface Sp

Date post:	02-Jan-2016
Category:	Documents
Upload:	belle1405
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