The administrator can use the CUCM Admin Panel to configure options for the account lockout.

To configure account lockout options, an administrator can perform either of the following:

1. Click the Locked by Administrator checkbox in the user credential page.

2. Set the number of login attempts, which signifies the number of failed logins due to invalid credentials.

3. Set the maximum idle time (in days) and if the user does not login for that many days, the account is locked.

In case of account lockout, JTAPI delivers detailed exceptions without any warning messages. JTAPI does not allow applications to modify any of these values, it only reports the information.

The Agent Greeting feature enables the JTAPI application to instruct the Cisco Unified Communications Manager to automatically play a pre-recorded announcement following a successful media connection to the agent device. The greeting helps to keep the agent sounding fresh as they do not have to repeat common phrases on each call. Agent Greeting is audible for the agent and the customer.

Agent Greeting can be initiated from any phone with a Built-in-Bridge (BIB). A call is initiated from the BIB to the DN specified in the request. Applications are responsible for answering this call and playing the media.

There are two types of calls:

• A basic call between the customer and agent.

• A secondary call, known as the Interactive Voice Response (IVR) call, which is created between an IVR device and the BIB of the agent phone.

The application invokes the new Agent Greeting API on a call, which creates an IVR call. The application then answers the call, and is responsible to play a recorded message.

The connection is not created for the agent on the IVR call, and as a result, the applications see the secondary call only. The IVR call has only one connection to play the IVR message.

Regardless of whether or not the application observes the IVR device, the Agent Greeting media plays. Observers on the agent receive an event to start the media. When the media finishes, the application must disconnect the IVR or CTI port that streams the media. When the second call is disconnected, an event is sent to observers on the agent and receives an event to end the media.

This feature is available only on phones that have BIBs. The majority of Cisco Unified IP Phones have BIBs, but the feature may not be available in various older or lower-end phone models. Administrators must enable the BIB for the device and configure it using the Cisco Unified Communications Manager Admin panel.

Whenever a request to addMediaStream is made, JTAPI blocks the request until the IVR device answers the call or CTI responds with a timeout error. Due to this, the JTAPI thread that invoked the addMediaStreamRequest cannot answer its own call, because it is blocked waiting for the request to finish.

Applications intending to use this feature must ensure that one of the following is applicable:

• The IVR DN is configured to auto-answer incoming calls

• A separate JTAPI thread or application is set up to answer on the IVR DN

  Interface changes

See CiscoTerminalConnection, CiscoFeatureReason, CiscoJtapiException, CiscoMediaStreamStartedEv, CiscoMediaStreamEndedEv

From release 10.5(2) Cisco Unified Communications Manager now supports the following encryption algorithm IDs:

• CiscoMediaEncryptionAlgorithmType

• CiscoMediaEncryptionAlgorithmType.AES_128_COUNTER_80

• CiscoMediaEncryptionAlgorithmType.F8_128_COUNTER_32

• CiscoMediaEncryptionAlgorithmType.F8_128_COUNTER_80

• CiscoMediaEncryptionAlgorithmType.AEAD_128_COUNTER

• CiscoMediaEncryptionAlgorithmType.AEAD_256_COUNTER

The CiscoMediaEncryptionAlgorithmType.AEAD_128_COUNTER and CiscoMediaEncryptionAlgorithmType.AEAD_256_COUNTER will be negotiated only for a secure call between two SIP endpoints.

CTI ports can register with any of the above algorithms, but will negotiate on AES_128_COUNTER_80 for secure calls.

Note	From Release 12.5(1)SU5 onwards, CTI ports can register with any of the above algorithms for secure calls. For more information, see "Stronger Cipher Suites on CTI Ports" section in Security Guide for Cisco Unified Communications Manager.

Certain IP phone types support an alternate language script other than the default script that corresponds to the phone-configurable locale. For example, the Japanese phone locale has two written scripts. Some phone types support only the default Katakana script, while other phones types support both the default script and the alternate Kanji script. Because applications can send text information to the phone for display purposes, they need to know what alternate script a phone supports, if any.

The new getAltScript() method provides alternate script information for an observed device. Currently there is only one known alternate script: Kanji for the Japanese locale.

JTAPI provides a new method for CiscoTerminal to provide alternate script information.

JTAPI exposes the API, CiscoTerminal.isBuiltInBridgeEnabled() to let applications know if the BIB capability is enabled on the terminal or not. Accordingly, the return value is true or false.

This API throws MethodNotSupportedException if it is invoked on a CiscoMediaTerminal or a CiscoRouteTerminal as these devices do not support a BIB.

This API throws InvalidStateException if invoked on a terminal that is not registered with the Cisco Unified Communications Manager.

This version of the Cisco Unified JTAPI supports the Arabic and Hebrew languages, which users may select during installation and in the Cisco Unified JTAPI Preferences user interface.

In order to support this feature for Linux based JTAPI client machines, auto updater feature has the following changes in its interface. The interface required that applications provide component name, provider IP address, user name and password. Applications do not need to specify an URL for downloading the component. This is done to avoid the issue with updater application in case URL changes between various releases of Cisco Unified Communications Manager Administration.

A new API called "Replace()" is part of the component interface. This facilitates replacing of old component with a newly downloaded component. The following section defines the operation of updater after the new interface changes. The new updater will:

• Use the same API signature as the old one.

• Create a file newjtapi.jar in the current folder of application which is the new version of the jar file.

• Copy the current jtapi.jar to a file by name component.temp in the classpath specified.

• Replace the current jar file with the new jar file. At the end of this operation, the current jar file becomes the component.temp and new jar file becomes jtapi.jar. Applications can still use old component interface which take URL either by specifying the URL themselves or by querying the URL through the new interface provided on CiscoProvider. The API required to get the URL information is present in the Interface summary for this feature. This operation is supported for both Unix and Windows.

  Backward compatibility

This feature is not backward compatible.

This feature provides applications with the ability to enable or disable AutoAccept for the addresses on CTIPorts and Route Points. When AutoAccept status changes for the address, Cisco Unified JTAPI provides the event to inform the application for changes.

Note	The maximum number of lines that are supported for route points equals 34.

The new interface setAutoAcceptStatus(), provided on the CiscoAddress object, allows the capability to set AutoAccept to ON or OFF. Interface getAutoAcceptStatus(), also provided on the CiscoAddress object, allows applications to query the current status of AutoAccept on the address.

When AutoAccept status changes for the address, applications get CiscoAddrAutoAcceptStatusChangedEv on AddressObservers. This event includes the interface getTerminal(), which returns the terminal on which the AutoAccept status gets changed, and the interface getAutoAcceptStatus(), which returns integers that specify whether AutoAccept is ON or OFF. If an address observer is not added, the event does not get provided.

The following interfaces support AutoAccept on CTIPort and RoutePoint:

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for AutoAccept on CTIPort and RoutePoint, see Message Sequence Charts

Be aware that when the Cisco Unified Communications Manager is upgraded to a higher version, the APIs may or may not be compatible with the new Cisco Unified Communications Manager version. Ensure that the APIs are upgraded to a compatible version, so the applications work as expected. Because the APIs are installed locally on the client server, the upgrade must take place on multiple machines. In the case of fewer client applications, you can easily do this by connecting to the Cisco Unified Communications Manager Administration and downloading and installing the Cisco Unified Communications Manager compatible plug-in.

For multiple client applications, this feature provides a facility by which an application at startup can identify itself to a web server via an HTTP request and receives a response with the version of the required JTAPI API.

The application compares the version that is available on the server to the local version in the application classpath and determines whether an upgrade is necessary. This allows applications to refresh the jtapi.jar component to match the Cisco Unified Communications Manager and provides a way to centrally deploy the jtapi.jar to which applications can auto update.

The API that is required to perform this functionality gets packaged in the form of an updater.jar. The jtapi.jar and updater.jar get packaged with the standard manifest, which can be used to compare versions.

Note	This feature does not update JTAPI Preferences, JTAPITestTools, Updater.jar and javadoc components. If applications require these components, install JTAPI from the Cisco Unified Communications Manager plug-in pages. Auto Update supports JTAPI Release 2.0 and later.

Refer to Cisco Unified JTAPI Installation for more information.

The following new or changed interfaces exist for autoupdate of APIs:

Class com.cisco.services.updater.ComponentUpdater

Interface com.cisco.services.updater.Component

The Autoupdater feature in JTAPI also allows applications to download the latest version of JTAPI.JAR directly from the Cisco Unified Communications Manager.

1. Updater creates a newjtapi.jar file in the current folder of the application, which represents the new version of the jar file that was downloaded from the Cisco Unified Communications Manager.

2. Updater copies the current jtapi.jar to a file that is named component.temp in the classpath specified.

3. Updater replaces the current jtapi.jar file with the new jtapi.jar file.

At the end of this operation, the current jar file becomes component.temp and the new jar file becomes jtapi.jar. This operation is supported for both Linux and Windows.

Example Usage of Autoupdater

Command Line : java com.cisco.services.updater.ComponentUpdater <server> <component name> <login> <passwd>Component localComponent,  downloadedComponent;
ComponentUpdater updater  =  new ComponentUpdater();
String localPath  =  updater.getLocalComponentPath(args[1]);
localComponent  =  updater.queryLocalComponentVersion("jtapi.jar",  localPath);
localComponent.copyTo("component.temp");
String provString  =  args[0] + ";login = " + args[2] + ";passwd = " + args[3];

CiscoJtapiPeer peer  =  (CiscoJtapiPeer) (JtapiPeerFactory.getJtapiPeer(null));
CiscoJtapiProperties tempProp  =  ( (CiscoJtapiPeerImpl) (peer)). getJtapiProperties();
tempProp.setLightWeightProvider(true);

Provider provider  =  peer.getProvider(provString);
String url  =  ( (CiscoProvider) (provider)).getJTAPIURL(); provider.shutdown();
Component serverComponent  =  updater.queryServerComponentVersion("jtapi.jar",  url);

downloadedComponent  =  serverComponent.fetchFromServer();
int retVal  =  downloadedComponent.replaces(localComponent);

The "replaces" API will replace the existing JTAPI version with the new version.

Note	The updater will only update the JTAPI.JAR file and not the other sample applications and Cisco JTAPI documentation that are bundled with the JTAPI plug-in. To get these other components, applications must download the plug-in from the Cisco Unified Communications Manager and install it.

The Barge Feature provides the ability for shared addresses to barge into an established call of address on another terminal. This feature gets activated when an address TerminalConnection is in the passive state and CallCtlTerminalConnection is in the bridged state. This version of Cisco Unified JTAPI only supports feature activation manually on application-controlled terminals (IP phones). For this release, you cannot activate the feature through an API.

The Privacy feature provides the ability to enable or disable other shared addresses to barge into call. When privacy is enabled, other shared addresses cannot barge into a call and vice versa. Privacy represents a terminals property. IP phones have a "Privacy" softkey and pressing it enables or disables the privacy. Privacy can be dynamically enabled or disabled for the active calls on the terminal. When privacy is on for the call, the TerminalConnection for the call appearances on the shared address appear in the "InUse" state. If privacy status changes during the CallProgress, CiscoTermConnPrivacyChangedEvent gets delivered to the application.

Two types of barge feature functionalities exist in Cisco Unified Communications Manager: one uses built-in conference bridge called "Barge, " while another uses shared conference bridge resources called "CBarge". From the application point of view, no interface changes exists between Barge and CBarge; however, some behavioral changes, which are described in the message flow diagram in Message Sequence Charts occur.

Barge, CBarge, and Privacy have these interfaces:

Interface CiscoTerminalConnection.getPrivacyStatus()

booleangetPrivacyStatus()

This interface returns the privacy status of a call on the terminal.

Interface CiscoTermConnPrivacyChangedEv

javax.telephony.TerminalConnectiongetTerminalConnection()

A new reason code, CiscoCall.CAUSE_BARGE gets added to CiscoCall for barge events.

JTAPI provides CallCtiCause as CiscoCall.CAUSE_BARGE when a SharedLine TerminalConnection or CallCtiTerminalConnection goes to an active or talking state as a result of barge. This cause code also gets provided in CallCtiEvents for dropping temporary calls that are created during the barge operation.

This cause code is not provided for the CBarge scenario.

For details on these interfaces, see Cisco Unified JTAPI Extensions To view the message flow for barge, CBarge, and privacy, see Message Sequence Charts.

The Call Control Discovery (CCD) feature facilitates provisioning for inter-call agent communications. It uses the Service Advertisement Framework (SAF) network service to advertise itself as a call control entity and to discover other call control entities (Cisco Unified Communications Managers or CMEs) on the network so that it can dynamically adapt their routing behavior.

When a call is made between two devices on different clusters and the call is rejected with a cause code other than unallocated , unassigned number and user busy, the CCD feature fails over the call to a PSTN network. That is, the call is routed through a PSTN network instead of an IP network to reach the same destination.

JTAPI supports the SAF CCD feature. However, applications are not notified when a normal SAF call fails over to a PSTN trunk.

JTAPI exposes a new reason CiscoFeatureReason.REASON_SAF_CCD_PSTN_FAILOVER for the new connection created for the redirect or forward destination. This occurs when there is a redirect or forward across the cluster through an SAF trunk and the call fails over to a PSTN trunk.

Cisco Unified JTAPI supports setting the Call Forward feature according to the JTAPI Specification. Cisco Unified JTAPI implementation does not support all the forwarding characteristics but supports only the FORWARD_ALL attribute for the Address. Applications can invoke setForwarding, getForwarding, and cancelForwarding methods on a CallControlAddress object, but the CallControlForwarding instruction can only be of type FORWARD_ALL.

This feature provides a mechanism to override the call forward all feature. If a user (CFA Initiator) sets CFA to another user (CFA target), the CFA should be ignored if the CFA target calls the CFA initiator. This would allow the CFA Target to reach the CFA Initiator for important calls.

The behavior of this CallManager feature is configurable via service parameter - CFADestinationOverride.

Example: Alice has a phone with DN 1000 * Bob has a phone with DN 2000 * Daniel has a phone with DN 4000 * Alice does a CFA to 2000

CFA behavior * Bob calls Alice. Call goes to Alice and does not follow CFA back to himself. * Daniel calls Alice. Call follows CFA to Bob. * Bob answers and transfers the call to Alice. Bob can do this because Alice has her phone forwarded to Bob. There is no interface change to JTAPI layer with this feature. However JTAPI applications could perceive a difference in behavior when CiscoAddress.setForward() API is invoked. In scenario where CFA target calls the CFA initiator as described in example, call is not forwarded if feature is enabled.

Cisco Unified JTAPI supports user interactions with Call Park and reports the appropriate events to the applications. When a call is parked from an IP phone, the connection that belongs to the parking address moves into Disconnected state, and the associated TerminalConnection moves into Dropped state. A new connection in queued state for the park number gets created.

If an application is monitoring only the address that parked the call, all existing connections get Disconnected, TerminalConnections get Dropped, and the call moves to Invalid state.

Call Pickup enables devices to receive alerts within Call Pickup Groups and events, to act on these alerts by invoking APIs that support variants of Call Pickup.

These APIs allow applications to gather information about existing Call Pickup groups, and register and unregister for receiving pickup alerts for specific pickup groups.

JTAPI supports invoking Pickup, Group Pickup, Other Pickup, and Directed Call Pickup from applications. In Cisco Unified Communications Manager releases prior to release 8.0(1), all these features except Other Pickup were supported as observed events, but were not invoked.

Note	Call Pickup is not supported on CTI route points.

Cisco Unified JTAPI sends CiscoTermConnSelectChangedEv event whenever the call is selected either by feature or by manually. Once application receives the event, application can use TerminalConnection.getSelectStatus() to get proper call select status. There are three possible statuses by calling TerminalConnection.getSelectStatus() as follows:

• CiscoTerminalConnection. CISCO_SELECTEDNONE: The select status means that the call is not selected

• CiscoTerminalConnection. CISCO_SELECTEDLOCAL: The select status means that the call is selected on the terminal connection

• CiscoTerminalConnection. CISCO_SELECTEDREMOTE: Passive TerminalConnection will get this select status if the call is selected by it's shared line

This feature enables applications to know whether the call is a normal call or a temporary call, when the CallFwdAll key is enabled.

JTAPI exposes this information through the API getCFwdAllKeyPressIndicator() which is exposed on the CiscoCall interface. This API enables the application to know if the call is created due to pressing of CallFwdAll softkey or not. The newly added getCFwdAllKeyPressIndicator()” could return following constants that are also new:

• If it is pressed on a phone that is in on-hook state to set CallFwdAll, this API returns CiscoCall.CFWD_ALL_SET.

• If it is pressed on a phone that is in on-hook state to clear the CallFwdAll, this API returns CiscoCall.CFWD_ALL_CLEAR.

• If the call is made first and then the user presses CallFwdAll key when phone is in off-hook state, this API returns CiscoCall.CFWD_ALL_NONE.

  Interface changes

See CiscoCall

You can select or unselect call on a phone for doing DirectTransfer or join or any other feature operation. When a SharedLine user selects a call, the RemoteInUse shares line TerminalConection will go passive, and CallCtlTermiCallConnection goes in InUse state. When call is unselected, CallCtlTerminalConnection goes into a bridged state. An application cannot invoke any API on Passive/InUse TerminalConnection. CallProcessing also performs a Select/UnSelect operation during features (such as transfer/conference) operation. Applications will also perceive these events if the applications monitor RemoteInUse terminal.

For example, if A and A' are SharedLine, and A selects the call, CallCtlTerminalConnection of A' goes into a passive or InUse state. If A "UnSelects" the call, the CallCtlTerminalConneciton of A' goes into the passive or bridged state.

To view the message flow for CallSelect or UnSelect, see Message Sequence Charts

Currently Cisco Unified JTAPI certificate download API has some security issues, to solve the problem, Cisco provides new certificate download APIs. New APIs require applications to specify a certificate pass phrase and the certificate pass phrase is used to encrypt Java key store where client/server certificates are stored.

Old certificate download APIs are deprecated, however, it will still remain for some time to avoid backward compatibility issue for applications. Cisco highly recommends to migrate the application to new APIs.

Cisco Unified JTAPI also provides new API deleteCertificate() and deleteSecurityPropertyForInstance() that can be used by application to delete certificates already installed. To change pass phrase for certificate java key store, the application must delete the old certificate by using this API and upload new certificate.

JTAPIPreferences UI security tab enhancement provides two new buttons, one for DeleteCertificate and another for Update Certificate. DeleteCertificate button allows users to delete the certificate for required username/instanceID. Update Certificate button allows users to upload the certificate from CAPF server. If certificate update is successful, certificate update box is updated to show Updated; authorization string and certificate pass phrase are cleared. If certificate update operation fails, certificate box continues to show status Not Updated status unless certificate was previously updated. User/Applications must provide certificate pass phrase every time they try to update certificate, Cisco Unified JTAPI does not save certificate pass phrase for security reason in any circumstances. Applications own the responsibility to secure the pass phrase and provide it through API when needed.

Currently, TSP hardcodes the DeviceTypeName depending on the DeviceType. When a new device type is added, we have to manually add the new device type name to the list of supported devices. Because CTI does not fetch and store the device type name in its cache, TSP cannot get this info from CTI. TSP needs to update the device type name when a new device type is added without any manual intervention.

In JTAPI, the changes have been made to ensure that QBE interface changes to handle the receive devicetypename that is sent from CTI and is stored in the deviceInfo structure. It is not used anywhere in JTAPI and will not be exposed to applications. Only the QBE interface changed as follows:

public DeviceRegisteredEvent ( String ride,  int deviceType,  boolean allowsRegistration,  int deviceID,  boolean loginAllowed,  UnicodeString userID,  boolean controlled,  int reasonInt,  int registrationType,  int unicodeEnabled,  int locale, 
// added for deviceTypeName change 
String devTypeName) {
public DeviceUnregisteredEvent ( String deviceName,  int deviceType,  boolean allowsRegistration,  int deviceID,  UnicodeString userID,  boolean controllableBool,  int reasonInt ,  int locale, 
//added for devtypename support
String devTypeName) {

In JTAPI, the terminal object represents the logical endpoint for a call and is presumed to be able to receive and transmit data (digital encoded voice samples, for example). Thus, terminals in JTAPI represent Cisco Unified IPPhones. Even though gateways terminate media, terminals do not represent them. The CiscoMediaTerminals in particular represent a special kind of endpoint for which applications take responsibility for media termination.

The following four steps associate with using CiscoMediaTerminals:

• Provisioning

• Registration

• Adding Observers

• Accepting Calls

Endpoints represent the entities within the Cisco Unified Communications Solutions platform that terminate media, such as IP telephones and gateways. A call from one endpoint to another results in media flowing between the two endpoints. All endpoints in the Cisco Unified Communications Solutions platform transmit voice data by using real-time protocol (RTP). The Cisco Unified Communications Solutions telephones and gateways, for example, include built-in RTP stacks. Applications may also act as endpoints in a Cisco Unified Communications Solutions system; that is, they may terminate media. Because all Cisco Unified Communications Solutions endpoints use RTP, applications also must be able to transmit and receive RTP packets.

If a Cisco Unified Communications Manager server fails, the associated devices re-home to the next Cisco Unified Communications Manager server in the group. The prioritized list of Cisco Unified Communications Managers in the device pool information configuration for each device defines this process.

Failure of a Cisco Unified Communications Manager server only results in a partial outage of devices in the cluster. Those devices remain available following a successful Cisco Unified Communications Manager failover and registration with a secondary Cisco Unified Communications Manager.

Note

A device such as a Cisco Unified IPPhone 7960 fails over to a secondary Cisco Unified Communications Manager server only when no active calls exist on that device. The failure of a Cisco Unified Communications Manager server during a call results only in termination of observation of that device. The media path continues to exist but without any further call control features.

Cisco Unified JTAPI communicates this partial outage to applications by using CiscoAddrOutOfServiceEv and CiscoTermOutOfServiceEv events. When the Cisco Unified Communications Manager fails over, the device must successfully register to the secondary Cisco Unified Communications Manager before the device is available to the JTAPI applications. Cisco Unified JTAPI will send the CiscoAddrInServiceEv and CiscoTermInServiceEv events.

The Provider remains in service during this time. Devices on other Cisco Unified Communications Manager servers remain available for call control. The events get sent on callbacks of the respective Address or Terminal observer objects. CiscoAddrOutOfServiceEv and CiscoAddrInServiceEv events get sent to an object that is implementing the AddressObserver and get added to an Address by using the addressChangedEvent() callback object method. The CiscoTermOutOfServiceEv and CiscoTermInServiceEv events get sent to an object that is implementing the TerminalObserver interface and get added to a Terminal that is using the terminalChangedEvent() callback method.

If the devices are currently in a call, a CallObservationEnded message is sent on the CallObserver callChangedEvent() callback, followed by the CiscoAddrOutOfServiceEv and CiscoTermOutOfServiceEv messages.

Note

Applications must monitor for and respond to the CiscoAddrOutOfServiceEv, CiscoTermOutOfServiceEv, CiscoAddrInServiceEv, and CiscoTermInServiceEv events before the calling call control functions on the address or terminal. Applications that do not support this action may encounter unexpected errors because the applications do not know the exact state of the system.

You can configure Cisco Unified IP 7931G phones in two modes:

• NoRollOver

• RollOver (across the same DN or across different DNs)

When Cisco Unified IP 7931G phones are configured in NoRollOver mode, they operate like regular phones that are running SCCP, and in this mode transfers or conferences cannot occur across the different addresses. JTAPI will support controlling and monitoring of a 7931G phone when it is configured in NoRollOver mode.

In RollOver mode, Cisco Unified IP 7931G phones support transfer or conference across different addresses. In this mode, JTAPI does not allow controlling and monitoring of a Cisco Unified IP 7931G phone. Applications see such terminal/addresses as restricted. If a Cisco Unified IP 7931G phone is in the control list of an application user and the phone configuration changes from NoRollOver to RollOver mode, JTAPI sends a CiscoAddrRestrictedEv event for addresses on the Cisco Unified IP 7931G phone and sends a CiscoTermRestrictedEv for terminals, with cause CiscoRestrictedEv.CAUSE_UNSUPPORTED_DEVICE_CONFIGURATION.

However, if the phone configuration changes from RollOver to NoRollOver mode, JTAPI sends a CiscoAddrActivatedEv event for addresses on the Cisco Unified IP 7931G phone and sends a CiscoTermActivatedEv for terminals.

If a Cisco Unified IP 7931G phone that is configured in RollOver mode transfers or conferences to JTAPI-controlled addresses, JTAPI applications do not recognize a common controller in the final and the consult call. This would provide different behavior to the JTAPI application. Depending on how the JTAPI application is processing information that is provided in events, applications may require changes to handle JTAPI events for this transfer or conference scenario.

You can disable transfers and conferences across the line by configuring the Cisco Unified IP 7931G phone to NoRollOver mode through the phone configuration window of Cisco Unified Communications Manager Administration.

There are two new cause codes for the CiscoRestrictedEv interface. When the terminal or address is restricted because a Cisco Unified IP 7931G phone is configured in RollOverMode, JTAPI sends CiscoAddrRestrictedEv with cause CiscoRestrictedEv.UNSUPPORTED_DEVICE_CONFIGURATION. This release also introduces a default cause code CAUSE_UNKNOWN, which applications should handle.

Cisco Unified JTAPI supports multiple languages for the JTAPI installation and user preference UI. When JTAPI launches, you receive options for choosing languages for the installation. After choosing a language, further installation instructions display in the chosen language. The first option always specifies English. If certain phrases are missing in the locale language, the instructions default to English. See Cisco Unified JTAPI Installation for more information.

Cisco Unified JTAPI supports control of analog phones that are connected to the Cisco VG248 and ATA 186 Analog Phone Gateways. By adding the Cisco VG248 and ATA 186 Analog Phone Gateways to the user-controlled list, applications can control the devices.

Applications receive events for the devices in a way similar to other IP phones. Applications can also initiate calls and invoke other features except answer Request through APIs. Make call works only when the device goes physically off hook.

Applications cannot answer calls from APIs for the devices. If an application attempts to answer () on TerminalConnection for the VG248 and ATA 186 Terminal, the system throws PlatformException with error CiscoJtapiException.COMMAND_NOT_IMPLEMENTED_ON_DEVICE. To answer calls, you must manually pick up the handset, and then you can invoke other call control features such as transfer, conference, blind transfer, and park from the API.

Cisco Unified JTAPI notifies the application of CTI-generated error codes. These codes return when an exception or error occurs in the CTIManager. The CTI returned error code propagates to the application separately. The application can extract the error code by invoking getErrorCode() method on the exception object, can get CTI error code name by invoking getErrorName() method, and can get error description by invoking method getErrorDescription().

CiscoJtapiExceptions interface defines error codes in JTAPI.

When a PlatformException is thrown, it can be queried to get the error code, which can be compared to the following.

CiscoProvAuthenticationInfoEv code returns to notify the application that the password is about to expire or has already expired. The application should have Provider Observer onto the Provider object to receive this event.

If the application invokes a connection and it fails because of an expired password, it will receive a PlatformException with a newly defined error code. For more information, see CiscoJtapiExceptions.

In the case of a failover, the application will not explicitly request a connection, and will not receive a PlatformException. As the provider will already have an observer, it will deliver a CiscoProvAuthenticationInfoEv to it with getDaysUntilPasswordExpiry() = CiscoProvAuthenticationInfoEv.PASSWORD_EXPIRED.

The following interfaces are enhanced to allow applications to get a CiscoRTPHandle from the events:

• CiscoRTPInputStartedEv

• CiscoRTPInputStoppedEv

• CiscoRTPOutputStartedEv

• CiscoRTPOutputStoppedEv

CiscoRTPHandle represents the callID of the call in Cisco Unified Communications Manager and stays the same as long as the call is active on the terminal. At any particular terminal/address, although the call and the associated GCID can change, CiscoRTPHandle will remain constant.

Prior to the JTAPI 2.0 release, Cisco Unified JTAPI applications did not have direct control over terminal events. Applications can now receive button pressed events by setting the appropriate filter in the terminal observer. Applications no longer need to add call observer to get RTP events.

When setButtonPressedEv gets enabled by using CiscoTermEvFilter, applications receive CiscoTermButtonPressedEv when a digit gets pressed on the phone.

The following new or changed interfaces exist for CiscoTerminal Filter and ButtonPressedEvents:

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for CiscoTerminal Filter and ButtonPressedEvents, see Message Sequence Charts

This event gets provided to the application when CiscoMediaTerminal or CiscoRouteTerminal registration fails asynchronously. Usually when registration fails, the application gets a CiscoRegistrationFailedException; however, it is possible that the registration request was successful, but the CTI rejected the registration. This event is provided for the cases where the registration request is successful, but the registration gets rejected. The application should have TerminalObserver to receive this event. Upon receipt of this event, the applications should reregister with the new parameter, depending on the error code that is provided for this event.

The following list provides the errors that get returned and the actions to take, by the application, to resolve them.

Wireless devices introduced by Cisco, for example the Cisco Cius, have the capability to move between WiFi networks and still retain their registration to a single CiscoUCM. However, due to the change in the network the IP address of the device might undergo a change.To indicate this change in IP address of wireless devices like Cius, Cisco JTAPI will expose a new interface to applications with the 9.0.1 release.

The new provider event - CiscoProvTerminalIPAddressChangedEv, will indicate that the IP address of the terminal has changed. Applications may choose to ignore this new event if they do not plan to support a Cius device.

On receiving this event, applications can query for the changed IP address of the terminal using the methods exposed in the new event or on the CiscoTerminal. This interface will also expose the IP addressing mode of the terminal, based on which IPv4/IPv6 address of the terminal can be queried.

public synchronized void providerChangedEvent( ProvEv[] eventList )
{
  try
  {
    for ( int i  =  0; i < eventList.length; i++ )
    {
    case (eventList[i].getID()) ){
        switch:
CiscoProvTerminalIPAddressChangedEv.ID:
      Terminal term  =  eventList[i]
                        .getTerminal();
        int ipAddrMode  =  eventList[i].getIPAddressingMode();
        InetAddr ipV4Addr  =  null;
        InetAddr ipV6Addr  =  null;
        if(ipAddrMode  =  CiscoTerminal.IP_ADDRESSING_MODE_IPv4)
        {
          ipV4Addr  =  eventList[i].getIPv4Address();
        }
        else if(ipAddrMode  =  CiscoTerminal.IP_ADDRESSING_MODE_IPv6)
        {
          ipV6Addr  =  eventList[i].getIPV6Address();
        }
        System.out.println(" TerminalName  = " + term.getName()  + 
           " ipAddressing Mode  = " + ipAddrMode +
           " IPv4 Address  = " + ipV4Addr +
           " IPv6 Address = " + ipV6Address);
      }
    }
    catch (exception e)
    {
      …
    }
  }

Cisco Unified JTAPI applications can clear phantom calls without dropping active calls. The CiscoAddress provides a clearCallConnections message to allow applications to clear the calls when no active calls exist on the Cisco Unified Communications Manager (formerly Cisco Unified Call Manager).

Click to conference feature provides interfaces on SIP trunk for applications such as Instant Messenger (IM) to add parties to a conference. Users can add other parties to a conference or remove parties by using such applications. When click to conference is used to add a party to conference, a call is offered to the target address. Only one connection for target address is created on this initial call. This call then gets added to conference which results in a new callID for the call on the target address and connections for other addresses in the call are created on the new call.

This section describes the interface changes in Cisco Unified JTAPI to handle the interactions when an address is added to a conference by using click to conference feature. When click to conference feature is used, consult call does not occur and Cisco Unified JTAPI applications do not receive CiscoConferenceStartEv or CiscoConferenceEndEv.

The feature can be disabled by turning off the "ENABLE CLICK TO CONFERENCE" CallManager service parameter.

The CTIManager provides a virtual representation of all the Cisco Unified Communications Managers in a cluster. Cisco Unified JTAPI applications communicate with the CTIManager instead of with a specific Cisco Unified Communications Managers. The CTIManager also maintains connection between Cisco Unified Communications Managers in a cluster. This allows a provider to represent any devices in the cluster under the CTIManager. Single-Box Configuration with JTAPI, Cisco Unified Communications Manager, and CTIManager in One Box illustrates “Single-Box Configuration with JTAPI, Cisco Unified Communications Manager, and CTIManager in One Box.” Redundant Cisco Unified Communications Manager and CTIManagers with JTAPI Deployed as a Separate Client illustrates “Redundant Cisco Unified Communications Manager and CTIManagers with JTAPI Deployed as a Separate Client.”

For more details about the cluster administration and device pool settings, refer to the Cisco Unified Communications Manager help information.

Note

In previous releases of Cisco Unified Communications Manager, applications that are running on Cisco Unified JTAPI could only control or monitor devices that are registered under a single Cisco Unified Communications Manager. If a Cisco Unified Communications Manager server went down, the connection between the Cisco Unified Communications Manager server and JTAPI would terminate and the Provider would shut down.

This mode helps to install JTAPI in systems that do not have GUI support (for example, a Linux account). The entire installation procedure is guided by a character input based menu, where the user is asked to provide a series of inputs, based on the install time conditions. This mode also provides all the other options provided by the GUI based installer.

The Component Updater interface is enhanced to allow applications to specify the location of updater log. Currently the updater log is created in the same directory as the application. This enhancement allows applications to specify the trace location.

When you conference two calls together, JTAPI specifies that all the parties from one call be moved to the other call. The call whose parties are moved away and that subsequently becomes invalid gets identified as the “merged” or “consult” call. The call to which the merged parties move gets identified as the “final” call hereafter. When parties move from the merged call to the final call, the application receives events that indicate that all parties dropped from the merged call and events that indicate that the parties reappeared on the final call.

To correlate the newly created connection objects with the old connection objects, use the CiscoConection.getConnectionID() method to obtain CiscoConnectionID objects for all old connections and all new connections. Matching connections will have identical CiscoConnectionID objects when you compare them by using the CiscoConnectionID.equals() method.

Conference support exists for the following methods:

• javax.telephony.callcontrol.CallControlCall.conference(Call)

• javax.telephony.callcontrol.CallControlCall.getConferenceController()

• javax.telephony.callcontrol.CallControlCall.getConferenceEnable()

• javax.telephony.callcontrol.CallControlCall.setConferenceController(TerminalConnection)

• javax.telephony.callcontrol.CallControlCall.setConferenceEnable(boolean)

Note	As of Cisco Unified Communications Manager Release 8.6, Cisco TelePresence MCU conference bridges are supported through JTAPI/TSP. From a JTAPI/TSP perspective, this conference bridge behaves in the same way as other supported conference bridges.

The Conference Feature provides the ability to conference more than two people into a single call. Events at CTI layer change, and Cisco Unified JTAPI gets enhanced to support the new CTI events.

Join Feature provides the ability to join multiple calls into one single conference call. This functionality now supports multiple calls. Applications need to pass an array of calls to be conferenced together.

The following new or changed interfaces exist for conference and joining of multiple calls into one conference call:

• The following interface allows Join to conference multiple calls into one conference call: Call.Conference(Call[] otherCalls)

  Note	A precondition requires that all the otherCalls must have controller as one leg of the call.

• The following new or modified interfaces exist in CiscoConferenceStartEv:

  • TerminalConnection getHeldConferenceController()—This interface proves useful only for the arbitrary conferencing of two calls and returns only one of the held calls.

  • TerminalConnection[] getHeldConferenceControllers()—This interface gets all of the held calls when multiple calls are joined.

  • TerminalConnection getTalkingConferenceController()—This interface returns the talking conference controller; however, if no talking conference controller exists when all the calls that are being joined into conference are held, this interface returns null.

  • Call getConferencedCall()—This interface returns only one of the many calls that are going to join into a conference and may not have any meaning for a join conference when more than two calls exist.

• New interface in CiscoConferenceEnded event Boolean isSuccess():

  This interface returns true or false depending on whether conference is successful or failed. Application can use interface to find whether conference is successful. The following events get defined as conference failure:

  • If the application issues the request Call.conference(otherCalls[]), this conference would be considered failed if one or more than one calls could join into conference. Applications can use the interface getFailedCalls() to find the failed call.

  • If no conference bridge is available and the conference could not complete at all, the application can use getFailedCalls() to get a list of calls that could not join the conference.

  • A party that was being conferenced dropped out before conference could complete.

• An interface on the CiscoConferenceEnded event (Call[] getFailedCalls()) gets all the calls that failed to join the conference when the conference fails.

The following new or changed behaviors exist for Conference:

• No hold or unHold message such as applications see when an arbitrary conference occurs.

• An arbitrary conference does not require, as a precondition, that any calls be in a talking state; however, all the otherCalls must have a controller as one leg of the call.

• Applications can conference two or more held calls into a conference call. In finalCall, the controller automatically gets retrieved to a talking state.

• Always include an active call in the request Call.Conference(otherCalls). If an active call is not included in the conference request, the request fails.

• If no active call exists at the controller, the Call.Conference(otherCalls) request remains successful; however, if one active call exists, it the request must include it.

• If the application does not have an active TerminalConnection that is passed as an argument, Call.consult() throws a PreConditionException/InvalidArgumentException.

• If the controller does not have an active TerminalConnection, Call.Conference()/Call.Conference(Call[]) throws a PreconditionException/InvalidArgumentException.

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for conference and join, see Message Sequence Charts

The conference chaining feature lets applications join two separate conference calls together. JTAPI applications see chained conference calls that are represented as two separate calls. When conference calls are chained, JTAPI creates a new connection for the conference chain and provides the CiscoConferenceChainAddedEv event on CallCtlCallObserver. When the conference chain is removed from the call, JTAPI disconnects the conference chain connection and provides the CiscoConferenceChainRemovedEv event on CallCtlCallObserver. From CiscoConferenceChainAdded/RemovedEv, applications can obtain CiscoConferenceChain, which provides a link for all the conference chain connections.

The following figure shows parties A, B, and C in conference call GC1 and parties C, D, and E in conference call GC2.

After the conference chain is created, the calls will look like the following figure.

Applications may get all the participants of a chained conference from the CiscoChainedConference object, which provides conference chain connections from all the conference calls that are chained together. By browsing through the connections list, applications can get a list of all the chained conference calls; however, applications must have at least one participant of each conference that is observed.

Note	For any conference scenario that involves chaining conferences or adding parties to a chained conference call, JTAPI will not provide ConferenceStarted/Ended event.

For more information, see the following topics:

• CiscoCall (for the getConferenceChain() interface)

• CiscoConferenceChain

• CiscoConferenceChainAddedEv

• CiscoConferenceChainRemovedEv

Applications can inform Cisco Unified Communications Manager that a consultation call for a transfer is being placed without establishing the media path. The system does not require establishing the media path for the intermediate call, if the consultation call is being placed to determine whether an agent is available before the actual transfer. The consultWithoutMedia method as defined in the CiscoConsultCall interface creates a consultation call without establishing the media path.

Note	The system allows only transferring of the consultation call; it does not allow the call to be in conference.

CTI Ports that are registered by an application include a mechanism similar to phone devices. When the Cisco Unified Communications Manager that is handling signaling for a CTIPort fails, the CTIManager recovers its services according to the device pool administration for this device. On a CTIManager failure, Cisco Unified JTAPI reregisters the CTI Port after it connects to the backup CTIManager. The CiscoAddrOutOfServiceEv and CiscoTermOutOfServiceEv events and the corresponding in-service events get sent after recovery of the CTI Port.

The application controls media streaming for these devices, and streaming continues even when the port is out of service. The application has responsibility to ensure that new calls do not get presented to the device until it is ready to accept them.

On a Cisco Unified Communications Manager server failure, the CTIManager recovers the device from the Cisco Unified Communications Manager server group as defined in the device pool administration for the CTI RoutePoint. When the primary Cisco Unified Communications Manager server recovers, the CTIManager attempts to recover the device on its primary Cisco Unified Communications Manager. This re-homing happens when no more calls exist on the device (similar to physical devices).

On a CTIManager failure, Cisco Unified JTAPI recovers the device on the backup CTIManager. The application receives notification of the availability of a device with the CiscoAddrOutOfServiceEv and CiscoAddrInServiceEv events.

Changes in personal device preferences and an increasing number of mobile and remote workers necessitates a more flexible Unified Communications solution that extends UC features with a Bring Your Own Device philosophy. Extend and Connect addresses this change.

Extend and Connect is a feature that allows administrators to rapidly deploy Unified Communications (UC) Computer Telephony Integration (CTI) applications which interoperate with any endpoint. Extend and Connect lets users leverage the benefits of UC applications from any location using any device. This feature allows interoperability between newer UC solutions and legacy systems, so customers can migrate over time as existing hardware is deprecated.

For more information, please refer to the Cisco Unified Communications Manager Features & Services guide.

The CTI Video Support feature allows the JTAPI Application to detect the multimedia capabilities of Line Devices; such as receiving video, sending video and both receiving and sending video. Cisco JTAPI provides the applications with the ability to expose the video capabilities of a terminal through the enhancement CTI Video Support. CTI applications can differentiate a video-enabled device from a non video-enabled device, and, a video call from an audio only call.

Cisco JTAPI provides a new API, getCiscoMultiMediaCapabilityInfo() on the CiscoTerminal to expose the multimedia capabilities of the device. Cisco JTAPI exposes the multimedia capabilities of the terminal after the device is in registered state. The multimedia capabilities of the terminal include:

• video capability (either none or video enabled),

• telepresence interoperability (either none or telepresence interoperability enabled on the device), and,

• screen count (to know the number of screens available on device).

The multimedia capabilities are exposed on a new interface CiscoMultiMediaCapabilityInfo, which has the following APIs to expose these capabilities.

• getVideoCapability(),

• getTelepresenceInfo(), and,

• getScreenCount().

The following APIs on the CiscoCall are used by the application to determine the calling party or called party multimedia capability prior to media setup.

• getCallingTerminalMultiMediaCapabilityInfo()—of the calling party in a call

• getCalledTerminalMultiMediaCapabilityInfo()—of the called party in a call

When the video capability of the terminal changes, a new Cisco JTAPI event, CiscoProvTerminalMultiMediaCapabilityChangedEv, notifies the application. This event is a JTAPI provider event, and is delivered when the application adds a Provider Observer. The terminal must be in registered state, to receive this event. Plugging in or plugging out the Cisco camera will not affect the video capability status, therefore, the event will not be triggered. However, you can modify the video capability using the Cisco UCM Administration Interface > Device Configuration page.

Note	The initial video capability API on CiscoTerminal is not supported for CTI Route Points and CTI Ports; however, they can receive the video information of the calling party.

The following devices supports the CTI Video feature:

• 89xx (SIP only)

• 99xx

• E20

• EX60/90

• CTS 500

• CTS 500-32

• Jabber(CSF)

• CTI RoutePoint

• CTI Port

A new CTIManager service parameter, IP Addressing Mode for Devices, has been added that allows you to configure the default CTI IP addressing mode for a device that does not have an associated Common Device Configuration.

When an application invokes the CiscoTerminal.getIPAddressingMode() method for a device that does not have a Common Device Configuration, JTAPI returns the value of the service parameter. The default setting for the new service parameter is IPv4 and IPv6. JTAPI communicates the value via CiscoTerminal.IP_ADDRESSING_MODE_IPV4_V6.

Note	For an individual CTI device, if that device has an associated Common Device Configuration, the IP Addressing Mode setting in the Common Device Configuration overrides the value of the IP Addressing Mode for Devices service parameter.

DeleteCall interface provides applications with the ability to delete a call that was created by using the createCall interface. This method accepts a call and throws an InvalidStateException if a provider is not in service or if the call is not in the IDLE state. DeleteCall moves the call to the INVALID state.

The following interface gets added to CiscoProvider:

{ public void deleteCall( Call call ) throws InvalidStateException;
}

Applications can use this interface to delete the call that was created by using createCall interface. This method accepts a call and throws an InvalidStateException if the provider is not in service or if the call is not in the IDLE state. DeleteCall moves the call to the INVALID state.

To successfully delete a call, the application creates the call by using createCall, and the call should be in the IDLE state.

Cisco Unified JTAPI supports automatic device recovery.

For devices such as the Cisco Unified IPPhone 7960, the re-homing feature represents part of the device firmware. On a primary Cisco Unified Communications Manager failure, the phone attempts to connect to the backup Cisco Unified Communications Manager when it is no longer on a call. This transition gets communicated to applications in the form of out-of-service and in-service events described in CTIManager Failure.

For virtual devices with no firmware such as CTI Ports and CTI RoutePoints, the CTIManager or Cisco Unified JTAPI performs the failover.

The Direct Transfer Across Lines feature allows support for connected transfer across lines. It allows two calls on different addresses of the same terminal to be transferred though the Transfer softkey on the phone or by using the transfer() API that is provided by Cisco Unified JTAPI. When a transfer is performed across lines, the JTAPI application behavior changes, as applications do not see a common controller address in final and consult calls. There is no change in the API and the same events are delivered whether calls are transferred on the same address (regular transfer) or across addresses (direct transfer across lines). This feature is supported on all supported phones, including CTI port, SCCP devices and SIP devices.

If an observer is not added to either of the two addresses from which the direct transfer is being attempted from the JTAPI API, then Cisco Unified JTAPI throws PlatformException with this error: Transfer controller is not set and could not find a suitable TerminalConnection.

This feature allows the user to park a call by transferring the call to a user-selected park code.

Applications require notification asynchronously of device additions or deletions from the user control list and device deletions from the Cisco Unified Communications Manager database. Applications also receive notification about line changes to a device. This notification gets sent to Cisco Unified JTAPI and propagates to applications with CiscoAddrCreatedEv, CiscoAddrRemovedEv, CiscoTermCreatedEv, and CiscoTermRemovedEv on the AddressObserver and TerminalObservers, respectively.

Note	Ensure that the device is registered for CTIPorts and CTIRoutePoints to receive the line change notification.

Do-Not-Disturb (DND) gives phone users the ability to go into the DND state on the phone when they are away from their phones or do not want to answer the incoming calls. The DND softkey enables and disables this feature.

From the user windows, users can configure the following settings for DND:

• DND Option-Ringer off

• DND Incoming Call Alert-beep only/flash only/disable

• DND Timer-value between 0-120 minutes. It specifies a "period in minutes to remind the user that DND is active".

• DND status-on/off

Note	The Application can only enable or disable the DND status.

• The Application can set the DND status by invoking a new interface on CiscoTerminal.

• JTAPI will also query the application about any change in the DND status when DND status is set by phone, Cisco Unified Communications Manager Administration, or application.

• The application must enable the filter in CiscoTermEvFilter to receive the preceding notification.

• The application can also query for the DND status through a new interface on CiscoTerminal.

• The application can also query for the DND option through a new interface on CiscoTerminal.

Note	This feature applies to phones and CTI ports. It does not apply to Route points.

In the case of emergency calls (made by a CER application) landing on an application that has DND enabled, the system overrides the DND settings and presents the call to the application. A new parameter, FeaturePriority, in the redirect() and selectRoute() APIs on CiscoCall, CiscoConnection, and CiscoRouteSession, respectively, makes this possible. The CER application that initiates the emergency call sets FeaturePriority as FeaturePriority_Emergency. The application sets the feature priority only for emergency calls. In the case of normal calls, applications either do not set the feature priority at all or set it to FeaturePriority_Normal. Applications do not set FeaturePriority_Emergency in case of normal calls. When initiating feature calls such as intercom, applications must specify FEATUREPRIORITY_URGENT.

Note	The connect() API on CiscoCall does not support the FeaturePriority parameter.

The application receives an exception if it tries to perform a getDNDStatus(), setDNDStatus(), or getDNDOption() before the device is in service.

A Post condition is added to DND to handle a DB update failure or device out-of-service situations if they occur after the setDNDStatus() request is sent. If a DB update failure or device out-of-service condition occurs after the setDNDStatus() request is sent, setDNDStatus() delivers a CiscoTermDNDStatusChangedEv to the application. If this event is not received, a post-condition time-out occurs, and the following exception is thrown: could not meet post conditions of setDNDStatus().

Do Not Disturb–Reject (DND–R) is an enhancement to the existing DND feature. Cisco Unified Communications Manager and JTAPI previously supported only the Ringer off DND. The user can reject calls with DND–Reject. You can set DND–R from the phone configuration window or the phone profile configuration window in Cisco Unified Communications Manager Administration.

When DND–R is enabled, the call is not presented to the terminal that has Call Reject enabled. There is no audible or visual indication of incoming calls on that end point. To enable DND–R, set the DND Status as true and the DND option to Call Reject.

FeaturePriority overrides DND. It can have any of the following values:

• 1: Normal

• 2: Urgent

• 3: Emergency

This release introduces FeaturePriority in connect() API on CiscoCall. FeaturePriority in selectRoute() and redirect() API is already supported in prior releases. When feature priority as EMERGENCY is specified in connect() API, and the destination terminal has DND–R enabled, the call still rings at the destination terminal and overrides the DND–R settings.

When a terminal has DND–R enabled and receives an intercom call, DND–R settings are overridden and call presents. This is because feature priority is always 2 (URGENT) for intercom calls.

In non- shared line scenario where A calls B and Terminal B has DND–R enabled, CallCtlConnFailedEv with cause USER_BUSY is delivered on A. Users would see the same behavior if DND–R is enabled on all the terminals that have shared DNs.

In the case of shared lines when at least one of the terminal does not have DND–R enabled and a call is placed to the shared line, Cisco Unified JTAPI delivers TermConnPassiveEv and CallCtlTermConnInUseEv for the terminals that have DND–R enabled (assuming the call was made with NORMAL feature priority). TermConnPassiveEv and CallCtlTermConnBridgedEv is delivered if DND–R is disabled on the terminal during a call.

A new event CiscoTermDNDOptionChangedEv will be sent to the terminal observer whenever the DND option changes on the phone window or Common Phone Profile window in Cisco Unified Communications Manager Administration.

Default DND option is Ringer–off and Route points do not support DND.

This feature provides the capability to drop any participants from a conference call. Cisco Unified JTAPI allows applications to drop participants from conference using the existing interface Connection.disconnect() even if the application is not observing the address for the connection. Previously, applications could only disconnect connections for which Address is an observed Address.

Feature behavior varies based on the settings for the Cisco Unified Communications Manager service parameter Advanced Ad Hoc Conference Enabled. If this service parameter is set to False, applications can drop connections for an unobserved address in a conference call only if the application observes the conference controller's address. If this parameter is set to True, applications can drop connections without any restriction.

Cisco Unified JTAPI provides an interface on CiscoConnection to get an array of CiscoPartyInfo objects for the connection. CiscoPartyInfo is used to disconnect participants from a conference using a new interface, disconnect(), provided on CiscoConnection. A normal line has only one CiscoPartyInfo on its connection, but a shared line has one CiscoPartyInfo for each line in the shared line. This enables applications to selectively disconnect a shared line participant if more than one shared line participants are in the conference call. Since shared line participants have only one connection, if the application uses the existing Connection.disconnect() API, it drops all the shared line participants.

Cisco Unified JTAPI provides an interface setDropAnyPartyEnabled() on CiscoJtapiProperties to enable or disable this feature and by default, it is enabled. Alternatively, applications can have the JTAPI ini parameter dropAnyPartyEnabled = 0 in jtapi.ini file to disable Drop Any Party feature and dropAnyPartyEnable = 1 to enable this feature. If dropAnyPartyEnable parameter is not present in jtapi.ini file, the feature is enabled by default.

Cisco Unified JTAPI also provides an interface, isConferenceCall(), on CiscoCall to determine if a call is a conference call. This simple method returns a Boolean.

This feature lets applications provide an IP address (ipAddress) and port number (portNumber) for each call or whenever media is established. To use this feature, applications must register the media terminal by supplying media capabilities. When a call is answered at this media terminal, CiscoMediaOpenLogicalChannelEv is sent to applications. This event gets sent whenever media is established. Applications must react to this event and specify the IP address and port number where media gets established.

A CiscoMediaTerminal represents a special kind of CiscoTerminal that allows applications to terminate RTP media streams. Unlike a CiscoTerminal, a CiscoMediaTerminal does not represent a physical telephony endpoint, which is observable and controllable in a third-party manner. Instead, a CiscoMediaTerminal represents a logical telephony endpoint, which may be associated with any application that terminates media. Such applications include voice messaging systems, interactive voice response (IVR), and softphones.

Note	Only CTIPorts appear as CiscoMediaTerminals through Cisco Unified JTAPI.

Terminating media comprises a two-step process. To terminate media for a particular terminal, an application adds an observer that implements the CiscoTerminalObserver interface by using the Terminal.addObserver method. Finally, the application registers its IP address and port number to which the terminal incoming RTP streams get directed by using the CiscoMediaTerminal.register method.

To register the ipAddress and portNumber dynamically on a per-call basis, applications must register by only providing capabilities that they support. Applications must react to CiscoMediaOpenLogicalChannelEv that gets sent whenever media is established. If any features are performed before applications react to CiscoMediaOpenLogicalChannelEv, the features may fail.

If the applications do not respond to this event during the time that is specified in the Media Exchange Timer in the Cisco Unified Communications Manager Administration windows, the call may fail.

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for Dynamic CTIPort Registration Per Call, see Message Sequence Charts

Note	The ChangeRTPDefaults interface is not supported on CiscoMediaTerminal.

The following new or changed interfaces exist for Dynamic CTIPort Registration Per Call:

The main purpose of this feature was intended to provide Location awareness for teleworkers and off premise users so that they can make emergency calls from off premises. The API's can also be used by all applications in a generic way as described below.

Primarily this feature adds two JTAPI API methods (selectRoute & redirect) that are overloaded to add an additional XML parmater to the list of their exisiting parameters. Any application can use these overloaded selectRoute() and redirect() methods to pass XML data to the call receiving side. The format of the XML data that can be passed is seen below:

<data>
     <item>
         <type>contact</type>
         <operation>append</operation>
         <protocol>SIP</protocol>
         <value>;+sip.instance = &quot;&lt;urn:uuid = *guid*&gt;&quot;</value>
     </item>
</data>

When an application sends XML data using one of the above API, CTI parses the data and extracts the text from the 'value' node in the XML and passes it on to CCM. CCM will then append this text to the outgoing SIP Invite message 'contact:' header. Once the end points like the SIP trunk or the SIP phone receives it, they can extract that data from the contact header and process it. Currently only SIP protocol's contact header field data is the only one supported but this can be expanded to include others headers fields and other protocols in future releases.

In the current release of CUCM only the following values for the XML nodes are supported: type: contact, operation: append, protocol: SIP. The value node in the above xml format is the one that carries the required application data to the end point.

The new parameter is a double byte array for overloaded selectRoute () Method to accommodate xml data for each selected routes and single byte array for the redirect () method. The parameter takes either a XML format String or a NULL value.

The CiscoAddress extension allows applications to set the status of the ringer for all lines on a device. No events generate when the ringer setting gets changed from the administration windows or anywhere else.

This feature enables the application to track any call uniquely. JTAPI associates a uniqueID with every Connection object. The same ID is exposed to the application through a new API getUniqueID(Terminal term) on the interface CiscoConnection. This uniqueID is only available for connection of observed addresses.

When a connection is created, the application can receive the uniqueID and write it in the Call Details Record. For Shared Line scenarios, each shared line has a uniqueID, which can be retrieved by passing the corresponding Terminal to getUniqueID API. UniqueID may or may not be the same for different shared lines depending on the scenario. The application can query the uniqueID corresponding to each shared line on receiving TermConnActiveEv for that shared line Terminal.

Whenever the uniqueID changes, JTAPI delivers CiscoConnectionUniqueIDChangedEv to the call observer of the application.

Note	As of Release 8.0(1), there is no supporting use case where JTAPI delivers CiscoConnectionUniqueIDChangedEv event to the application.

This feature allows the phone to participate in an EnergyWise enabled system. The phone reports its power usage to a EnergyWise compliant switch to allow the tracking and control of power within the customer premise. The phone provides alternate reduced power modes including an extremely low, off mode. The Cisco Unified Communications Manager administrator configures and exclusively manages the phones power state through vendor specific configuration on the Cisco Unified CM Admin pages.

When the phone turns off power after negotiation with an EnergyWise switch, it unregisters from Cisco Unified CM and enters Deep Sleep/PowerSavePlus mode. Phones automatically re-register back with the Cisco Unified CM once the Deep Sleep mode configured PowerON time is reached.

However, you can press the ‘select’ key on the Cisco Unified IP Phones Series 9900 and 6900 while in Deep Sleep/PowerSavePlus mode to wake up the phone, these phones automatically power on and re-register back with the Cisco Unified CM. However, for Cisco Unified IP Phones 7900 Series phones, you can neither power on nor re-register back with the Cisco Unified CM during Deep Sleep/PowerSavePlus mode unless the ‘PowerON’ time is reached. You can configure Deep Sleep mode on the Device page of the Cisco Unified CM. Configure Deep Sleep mode for the phones at least 10 minutes before the actual power off time to allow the information to synchronize between the switch and the phone.

The Power off idle timer enables only in the case when there is physical interaction on the phone. For example if there is a call on the EnergyWise configured phone during the deep sleep time and the user tries to disconnect the call from the application, then the power off idle timer defaults to 10 minutes but if the user disconnects the call manually from the phone, then the power off idle timer takes the value configured on the Cisco Unified CM device page.

When a terminal unregisters from Cisco Unified CM, JTAPI exposes CiscoProvTerminalUnRegisteredEV event to application with a new reason "CiscoProvterminal UnRegisteredEV.ENERGYWISE_POWER_SAVE_PLUS".

JTAPI sends CiscoTermOutOfServiceEv event to the application with the cause "CiscoOutOfServiceEv.CAUSE_ENERGYWISE_POWER_SAVE_PLUS" when a terminal goes out of service due to Deep Sleep mode configured.

JTAPI sends CiscoAddrOutOfServiceEv event to application with a new cause "CiscoOutOfServiceEv.CAUSE_ENERGYWISE_POWER_SAVE_PLUS" when an address goes out of service due to Deep Sleep mode configured.

Sample Code:public class MyTermObserver implements ProviderObserver {

  public void providerChangedEvent (ProvEv[] evlist) {
        for(int i = 0; evlist ! =  null && i < evlist.length; i++){
…
…
If ( evlisth[i] instanceof  CiscoProvTerminalUnregisteredEv){
CiscoProvTerminalUnregisteredEv ev  =  (CiscoProvTerminalUnregisteredEv)evlist[i];
if(ev.getReason()  =  =  
CiscoProvTerminalUnregisteredEv.ENERGYWISE_POWER_SAVE_MODE){
System.out.println(“Terminal Unregistered from CUCM because of deep
with the reason as ENERGYWISE_POWER_SAVE_PLUS
 ”);
}

This feature allows users to log in to an IP phone registered to a cluster with user profiles configured with another cluster. The Extension Mobility feature allows a user to log in to an IP phone to appear as user's desk phone temporarily, subject to the administrative policy. After logging on to an IP phone, the user can receive incoming calls normally routed to the user's desk phone and retain the personalized configuration, such as speed dials, services links and other user-specific properties.

Currently, Extension Mobility service is limited to a single Cisco Unified Communications Manager (Cisco Unified Communications Manager) cluster. A user provisioned in one cluster today cannot log in to an IP phone of another cluster with the Extension Mobility feature, even though both clusters may belong to the same enterprise. This limitation is overcome with the introduction of this new feature, which allows the user provisioned in one cluster to log in to an IP phone of another cluster.

With the existing behavior, when a user logs in to a terminal with a user ID that matches the user ID used by Cisco Unified Communications Manager JTAPI application, the terminal is treated as part of the control list and application is able to add call observer on the terminal and/or address.

As part this feature support, Extension Mobility profiles can be added to the user's control list via the Cisco Unified Communications Manager Admin pages. When a user uses Extension Mobily to log into a device using a profile in the control list, JTAPI delivers CiscoAddrCreatedEv and CiscoTermCreatedEv, and application can add call observer to control the terminal or address.

JTAPI exposes getCiscoCause () API on all provider events. For provider events associated with non-Extension Mobility login or logout scenarios, the cause delivered will be CiscoProvEv.NORMAL. For provider events associated with Extension Mobility login or logout scenario, the cause may be any of the below depending on the type of Extension Mobility login or logout:

CiscoProvEv.CAUSE_EM_LOGIN                    CiscoProvEv.CAUSE_EM_LOGOUT        CiscoProvEv.CAUSE_EM_LOGIN_PROFILE_ADD CiscoProvEv.CAUSE_EM_LOGOUT_PROFILE_REMOVE 

Interface changes explain more about each of these causes.

The following is a complete set of provider events that have API getCiscoCause():

CiscoAddrActivatedEvCiscoAddrActivatedOnTerminalEv
CiscoProvFeatureEv
CiscoProvTerminalCapabilityChangedEv
CiscoAddrRestrictedEv
CiscoTermActivatedEv
CiscoTermRestrictedEv
CiscoAddrCreatedEv
CiscoTermCreatedEv
CiscoAddrRemovedEv
CiscoTermRemovedEv
CiscoAddrAddedToTerminalEv
CiscoAddrRemovedFromTerminalEv.

JTAPI exposes getLoginType () on CiscoTerminal to indicate if the terminal is part of the Home or Visiting cluster when a user does an Extension Mobility login or logout. Accordingly, return value will be CiscoTerminal.NO_LOGIN, CiscoTerminal.NATIVE_LOGIN or CiscoTerminal.VISITOR_LOGIN.

Home Cluster is the Cisco Unified Communications Manager cluster from which the traveling EMCC user starts. This is the user's home cluster where the user profile resides.

Visiting Cluster is the Cisco Unified Communications Manager cluster which the traveling EMCC user visits. This is also the cluster that owns the phone at which the user does Extension Mobility login.

The Extension Mobility Login Username enables applications to get the Extension Mobility login username from the API provided on CiscoTerminal.

External Call Control enables Cisco Unified Call Manager (Cisco Unified Communications Manager) to route calls based on enterprise policies and presence-based routing rules of individual users. When call intercept is enabled, Cisco Unified Communications Manager queries the designated web services hosting the enterprise policies or user rules and routes the calls following the routing decisions returned.

Starting from Release 8.0(1), JTAPI supports wildcard routepoins, as well as translation patterns.

Cisco Unified Communications Manager generates a unique session identifier for each leg in a call. This feature enables the application to track a call end to end uniquely with a Session ID.

The methods accept a TerminalConnection object associated to that connection as a parameter, and return a String representing the UUIDs of the local and peer participant in a given CiscoConnection respectively. If a null object is passed as a parameter, the methods will return the UUID of the active TerminalConnection in the CiscoConnection.

device=<localUUID>;remote=<peerUUID>;

The Session ID is generated within Cisco Unified Communication Manager for non-SIP devices. SIP devices generate their own Session IDs and publish them in the SIP INVITE message to Cisco Unified Communication Manager. This information is visible to the application through the respective interface.

The Local Universal Unique Identifer (localUUID) for a CiscoConnection in a CiscoCall is generated in the peerUUID on the other side in the CiscoCall and vice versa.

If the Cisco call is shared between multiple devices, the CiscoConnection.getPeerUUID(null) on the calling side will return the UUID of any of the available terminals while the CiscoConnection on the called side is in CiscoConnection.ALERTING state. Once the call is answered CiscoConnection.getPeerUUID(null) on the calling side will return the uuid of the active TerminalConnection.

This feature allows Unified Communications Manager to operate in Federal Information Processing Standard (FIPS) mode. FIPS specifies a minimum security level for cryptographic functions, limitations on how data is stored, and which algorithms are allowed to be used to encrypt sensitive information. These strictly defined requirements are important to government agencies, hospitals, and other customers who're interested in a higher level of security.

To enable FIPS Compliance, Unified Communications Manager applications must request this mode when they download certificates with JTAPI and open a provider. When operating in FIPS, Unified Communications Manager experiences minimal performance loss. You can only witness this loss during the certificate downloads and opening of a Cisco Unified Communications Manager JTAPI provider. FIPS does not affect anything when the application is running.

Starting from Release 8.6(1), JTAPI can be configured as FIPS-compliant.

Important Notes

In FIPS, there are two distinct "cryptographic entities": The JTAPI application and the Unified CM server (or cluster). The FIPS compliance of one does not, in any way, affect the other. Setting JTAPI to run in FIPS compliance encrypts the client-side certificates with a FIPS-compliant algorithm, and connect using only approved SSL/TLS algorithms. It will not make the Unified CM server or cluster secure or FIPS-compliant.

Likewise, having the Unified CM operate in FIPS mode will not make JTAPI store certificates with FIPS-compliant algorithms. They are distinct items, separated by a”cryptographic boundary.”

Also, even if JTAPI operates in FIPS-compliant mode, your application may not. Your applications must handle cryptographic information and other sensitive data with special attention to be FIPS-compliant.

As mentioned earlier, applications that need to use FIPS compliance must not only explicitly request it, but also download cryptographic libraries and modify their classpath variables to include them.

Until Unified Communications Manager Release 12.5(1), JTAPI used RSA libraries for FIPS-compliant operations. With Release 12.5(1) and later, JTAPI on Windows uses RSA libraries, while on Linux it uses CiscoJ libraries.

As of Unified Communications Manager Release 14SU2, JTAPI uses BCFIPS libraries for all security-related operations. If configured to operate in FIPs mode, JTAPI moves BCFIPS libraries to approved only mode to enforce FIPS compliance.

The libraries are detailed below:

The RSA libraries are:

"jcmFIPS.jar" "cryptojcommon.jar", " cryptojce.jar" and "sslj.jar", are FIPS-compliant libraries from RSA, Inc.

The CiscoJ libraries are:

The CiscoJ libraries are "CiscoJCEProvider.jar", "log4j-1.2.17.jar", "slf4j-api-1.7.24.jar", "slf4j-log4j12-1.7.24.jar", "slf4j-simple-1.7.24.jar", "bcpkix-jdk15on-154.jar", and "bcprov-jdk15on-154.jar".

Note	From Release 12.5(1)SU5 on this train and up to 14SU1, "bcpkix-jdk15on.jar" and "bcprov-jdk15on.jar" are used instead of "bcpkix-jdk15on-154.jar" and "bcprov-jdk15on-154.jar" respectively.

Note	From Release 14SU2 to 15SU1, the BCFIPS libraries are: bc-fips.jar (version 1.0.2.3) bctls-fips.jar (version 1.0.12.3) bcpkix-fips.jar (version 1.0.5)

Note

From Release 15SU2 onwards, the BCFIPS libraries are: bc-fips.jar (version 2.0.0) bctls-fips.jar (version 2.0.19) bcpkix-fips.jar (version 2.0.6) bcutil-fips.jar (version 2.0.1) If you're upgrading the BCFIPS jars in your runtime environment to the latest supported (2.x as aforementioned) and an application is using an older jtapi.jar (earlier than 15SU2), while setting up the application environment, you must explicitly set "org.bouncycastle.jsse.fips.allowRSAKeyExchange" to True as a runtime system property to ensure that CAPF operation succeeds.

These libraries contain special implementations of several key cryptographic functions that replace the older implementation in jtapi.jar.

If your application contains a lib folder where third-party libraries are stored, the classpath looks as follows:

• For the JTAPI plugin using RSA libraries (refer above for library usage information as per the Unified Communications Manager release):

  ./libs/jcmFIPS.jar;./libs/cryptojcommon.jar;./libs/cryptojce.jar;./libs/sslj.jar;./libs/jtapi.jar

• For the JTAPI plugin using CiscoJ libraries (refer above for library usage information as per the Unified Communications Manager release):

  ./libs/CiscoJCEProvider.jar;./libs/CiscoJUtils.jar;./libs/CiscoJCEJNI.so;./libs/libssl.so;

  ./libs/libssl.so.1.0.1;./libs/log4j-1.2.17.jar;./libs/libciscosafec.so;./libs/libciscosafec.so.4;

  ./libs/libciscosafec.so.4.0.1;./libs/libcrypto.so;./libs/libcrypto.so.1.0.1;

  ./libs/slf4j-api-1.7.24.jar;./libs/slf4j-log4j12-1.7.24.jar;./libs/slf4j-simple-1.7.24.jar;

  ./libs/bcpkix-jdk15on.jar;./libs/bcprov-jdk15on.jar;./libs/jtapi.jar

• For the JTAPI plugin using BCFIPS libraries (refer above for library usage information as per the Unified Communications Manager release):

  ./libs/bc-fips.jar;./libs/bcpkix-fips.jar;./libs/bctls-fips.jar;./libs/jtapi.jar

Even with the classpath set this way, the JTAPI security code works the same way it does now unless the application specifically requests to run in FIPS mode.

To request that JTAPI run in a FIPS-compliant mode, applications must use some of the new methods that are introduced as part of this feature development and specify the new "fipsCompliant" parameter as True. For more information, see the following "Interface Changes" section.

Forced Authorization Codes (FACs) force the user to enter a valid authorization code prior to extending calls to specified classes of dialed numbers (DN), such as external, toll, or international calls. Authorization information is written to the Cisco Unified Communications Manager CDR database.

Client Matter Codes (CMCs) let the user enter a code before extending a call. Customers can use Client Matter Codes for assigning accounting or billing codes to calls that are placed, and Client Matter Code information is written to the Cisco Unified Communications Manager CDR database.

This feature enhances the forwarding logic to handle the No Bandwidth & Unregistered DN cases:

• No Bandwidth: When a call cannot be delivered to a remote destination due to no bandwidth, the system reroutes the call to the AAR Destination Mask or voice mail. The user makes these configuration changes from the directory number window of the Cisco Unified Communications Manager GUI.

• Unregistered DN: When a call is placed to an unregistered DN, the system delivers the call to a DN that is configured for Call Forward on No Answer (CFNA).

When a call is forwarded due to Call Forward No Bandwidth (CFNB) to another cluster destination over a trunk/gateway that is using QSIG, call history might get lost. For example, if Phone A calls Phone B, which is in a low bandwidth location, with CFNB set to forward calls to Phone C, which is in a different cluster, and the QSIG protocol is used for this intercluster forwarding, then the original called party and the last redirecting party might not get passed to the destination party.

GetCallID provides an interface on RTP events to access any call information, such as calling party or called party, so applications can link RTP events with the calls.

The callLegID that is received in the RTP events from CTIManager gets used to determine the ICCNCall on the client side. This call passes on to the JTAPI layer, and the CiscoCall gets determined, from which CiscoCallID is obtained. This information gets used to construct the RTP events that are delivered to the application.

The following interface gets added to CiscoRTPInputStoppedEv, CiscoRTPInputStartedEv, CiscoRTPOutputStoppedEv, and CiscoRTPOutputStartedEv:

{ public CiscoCallID getCallID();

}

GetCallInfo interface on address provides applications with the ability to query CallInfo on an address. A query returns the CiscoAddressCallInfo object, which contains information about the number of active or held calls, maximum number of active or held calls, and the call object for current calls on the address. This interface also specifies what calls are at a specific address at a specific time.

Use the following interface to get information about calls that are present at the terminal:

{ public CiscoAddressCallInfo getAddressCallInfo(Terminal iterminal);}

GetGlobalCallID provides an interface on the CiscoCallID to get the nodeID and the Global Call ID (GCID) of the call; this exposes the GCID information that is available in the internal call object.

The following methods get added to the CiscoCallID interface:

{    /**
     * returns the callmanager nodeID of the call
     */
public int getCallManagerID();

     /**
     * returns the GlobalCallID of the call
     */
public int getGlobalCallID ();
}

}

A hairpin call happens when the call leaves one cluster to some other device across the gateway, then comes back to a device in the same cluster. The GCID for the call coming back into the cluster would differ from the GCID that originally initiated the call, even though both are in the same cluster. In previous releases, if JTAPI controlled both parties, there were two connections: one for CiscoAddress.Internal and the other for CiscoAddress.External.

JTAPI supports hairpin calls when an application monitors both ends of the hairpin call. Previously, only one end of the hairpin call could be monitored because the address was represented only as a DN.

In the current release, if two addresses exist with the same DN but one is within the same cluster and the other is across the gateway, JTAPI creates a separate address object for the external DN, and only one connection is returned for an address, based on its type. This process avoids hairpin issues, as in previous releases when the address was represented only as a DN and when an application retrieved connections for the address it used to get two connections.

Since fixing these issues could have caused compatibility issues with previous releases, a generic solution for these issues was developed in this release. Calls that involve an external party with the same DN as the monitored local party are now properly supported; however, no new interface is added for this feature.

Currently JTAPI media events CiscoRTPInputStarted, CiscoRTPOutputStarted, CiscoRTPInputStopped and CiscoRTPOutputStopped do not indicate whether media is half duplex (receive only / transmit only) or full duplex (both receive and transmit).

This enhancement adds the capability to provide this information in a JTAPI media event. JTAPI provides an interface on the above media events to query whether media is half duplex or full duplex.

The half duplex media support feature does not impact JTAPI backward compatibility.

A new interface getMediaConnectionMode() is added to Cisco Unified JTAPI RTP events. This interface will return the following values depending on the media:

• CiscoMediaConnectionMode.NONE

• CiscoMediaConnectionMode.RECEIVE_ONLY

• CiscoMediaConnectionMode.TRANSMIT_ONLY

• CiscoMediConnectionMode.TRANSMIT_AND_RECEIVE.

CiscoRTPInputStarted/StoppedEv should only return RECEIVE_ONLY and TRANSMIT_AND_RECEIVE. The interface should not return NONE or TRANSMIT_ONLY. Ifthat happens, applications should ignore the event or log an error.

CiscoRTPOutputStarted/StoppedEv should only return TRANSMIT_ONLY and TRANSMIT_AND_RECEIVE. The interface should not return values NONE or RECEIVE_ONLY. Ifthat happens, applications should ignore the event or log an error.

CiscoMediaOpenLogicalChannedEv should only return RECEIVE_ONLY and TRANSMIT_AND_RECEIVE. The interface should not return values NONE or TRANSMIT_ONLY. Ifthat happens, applications should ignore the event or log an error.

public interface CiscoRTPInputStartedEv

public interface CiscoRTPOutputStartedEv

public interface CiscoRTPInputStoppedEv

public interface CiscoRTPOutputStoppedEv

The Hold Reversion feature provides applications with a notification when Cisco Unified Communications Manager notifies an address about the presence of a held call, when the call has been ONHOLD for a certain amount of time. Applications receive this notification as the CiscoCallCtlTermConnHeldReversionEv call control terminal connection event on their call observers on the particular address that has put the call ONHOLD. This notification is provided only once for the applications for the held call.

The event is sent only to the terminal connection of the terminal where the call was put on hold. If the address represents a shared line address, other terminal connections of the shared line address will not receive the event.

To receive this event, applications must add a call observer to the address. The cause for this event will be CAUSE_NORMAL. If the call observer is added after the hold reversion timer has expired and the notification has already been sent to the phone, applications will receive CiscoCallCtlTermConnHeldReversionEv with cause CAUSE_SNAPSHOT.

For more information, see CiscoCallCtlTermConnHeldReversionEv.

This feature enables the JTAPI application to observe addresses and terminals that are HuntList LineGroup members. Calls can arrive at these addresses either by another address calling it directly or through HuntPilot. When a call is made to HuntPilot, JTAPI creates a CiscoHuntConnection to represent HuntPilot and provides a Call Model that gives applications the information that the call is routed through HuntPilot. When a call is routed through HuntPilot and is connected to LineGroup Member, JTAPI call has three connections, two regular connections for calling and called addresses, and one CiscoConnection to HuntPilot through which that call was routed.

HuntPilot is not an observable address. The address representing Hunt Pilot is created when a call is made to a Hunt Pilot and is removed when the call is over. Applications cannot receive the Hunt Pilot address from the provider by using the getAddress() method.

In normal Hunt List calls, there are three connections, Calling, Hunt Pilot, and Hunt Member. When a call is made to the Hunt Pilot Directory Number, the call is offered to one of its members depending on the algorithm. The initial state of the call is Offering at the member. If members are not observed, the connection to Hunt Pilot goes through the normal states as CallCtrlConnection or Connection. If members are observed, connection to member goes to Offering state and the connection to Hunt Pilot goes to Established state. Applications must use the states of the observed party to track the state of the call.

call.getCurrentCalledParty() for a call to Hunt Pilot returns an address of type CiscoAddress.HUNT_PILOT. If the Hunt List member is the called address and is not observed by the application, connection to the member is created only when the call is answered.

CiscoHuntConnection extends CallControlConnection and can get into states that a call control connection could transition to expect for the network states.

Hunt pilots are represented by CiscoAddress objects and getType () would return CiscoAddress.HUNT_PILOT.

Only addresses returned for Cisocall.getCurrentCalledAddres () and CiscoCall.getCurrentCallingAddress () will have CiscoAddress.HUNT_PILOT type.

When calls to hunt pilot are involved in transfer or conference operations CiscoTransferStartEv, CiscoTransferEndEv, CiscoConferenceStartEv and CicsoConferenceEndEv are not delivered. Applications should use CiscoCallChangedEv to identify surviving call.

If consult calls or final call have CiscoHuntConnection, the application should not expect Transfer or Conference start and end events.

When configured in broadcast mode, all Hunt List members ring simulatenously. In JTAPI, call connections and terminal connections for Hunt List members are created only for members observed by the application.

Applications must enable this feature using the setHuntListFeatureEnabled (boolean) on CiscoJtapiProperties. This feature is disabled by default and applications are encouraged to adapt to the above call model and enable the feature using setHuntListFeatureEnabled () API. Observing a hunt list member without enabling the feature using setHuntListFeatureEnabled () is not a supported configuration and if observed, results in inconsistent call model and events. setHuntListFeatureEnabled() is introduced to enable applications that are currently using unsupported call scenarios with Hunt List to migrate to a supported model without breaking the existing functionality.

Applications can also enable this feature by adding, HuntListEnabled = 1, to jtapi.ini file and restarting the application.

In Cisco Unified CM 9.0, the support for hunt pilots is enhanced to expose the connected number as the modifiedCalledAddress in a call involving a hunt pilot.

With this enhancement, when a user calls a hunt pilot and the call is answered by the hunt member L1, call.getModifiedAddress() returns the address of the member L1, whereas call.getCurrentCalledAddress() returns the address of hunt pilot. Before the call is answered, both these values will return the address of hunt pilot.

The Intercom feature allows one user to call another user and have the call answered automatically with one-way media from the caller to the called party, regardless of whether the called party is busy or idle. The called user can press the talk back softkey (unmarked key) on their phone display, or the called user can invoke the join() JTAPI API, that is provided on TerminalConnection, to start talking to the caller. Only a specially configured intercom address on the phone can initiate an intercom call. Cisco Unified JTAPI creates a new type of address object named CiscoIntercomAddress for intercom addresses that are configured on the phone. The application can get all the CiscoIntercomAddresses that are present in the provider domain by calling the interface getIntercomAddresses () on CiscoProvider.

An intercom call can be initiated from the Cisco Unified JTAPI interface by calling the CiscoIntercomAddress.ConnectIntercom () interface. The application provides an intercom target DN for this interface. If the intercom target DN is preconfigured or preset by the application, the application can get the target DN by calling the CiscoIntercomAddress.getTargetDN() interface; otherwise, the application must provide a valid intercom target for the call to be successful.

An intercom call is autoanswered at the intercom target; Cisco Unified JTAPI will move TerminalConnection/CallCtlTerminalConnection at the intercom target to the Passive/Bridged state. The application can invoke a join () interface on the TerminalConnection of the intercom target to initiate talk back. If join () is successful, the TerminalConnection/CallCtlTerminalConnection of the intercom target will move to an Active/Talking state. For an intercom call, Cisco Unified JTAPI only supports the following interfaces:

• Call.drop ()

• Connection.disconnect ()

• CallCtlTerminalConnection.join ()

The application cannot perform any feature operations on an intercom call. Cisco Unified JTAPI will throw an exception if the application invokes redirect, consult, transfer, conference, or park for a Connection on a CiscoIntercomAddress. The application will also receive an exception if it tries to invoke setForwarding (), getForwarding (), cancelForwarding (), unPark (), setRingerStatus (), setMessageWaiting (), getMessageWaiting (), setAutoAcceptStatus (), or getAutoAcceptStatus () on CiscoIntercomAddress.

Applications can get the value of a configured intercom target DN and the label on a CiscoIntercomAddress from the provided API. Cisco Unified JTAPI provides two types of APIs: one to return the default and another to return the current value set for the intercom target. The default value is the intercom target DN and label that are preconfigured through Cisco Unified Communications Manager Administration. The current value is the interim target DN and label that the application sets. If the application has not set any value, the current value remains the same as the default value. Applications can invoke the API setIntercomTarget () on CiscoIntercomAddress to set the intercom target DN, label, and unicode label. Only one application can set the intercom target, label, and unicode label for an intercom address. If two applications try to set the value, the first succeeds, and the second receives an exception. When a intercom target DN and label changes, Cisco Unified JTAPI provides a CiscoAddressIntercomInfoChangedEv to the AddressObserver that is added to CiscoIntercomAddress. If the application has set an intercom target DN and label, and a JTAPI or CTI failover or failback occurs, JTAPI or CTI will restore the previously set value of the intercom target DN, label, and unicode label. If the JTAPI or CTI cannot restore the intercom target DN, label, or unicode label, Cisco Unified JTAPI provides a CiscoAddrIntercomInfoRestorationFailedEv to the AddressObserver on CiscoIntercomAddress. In the case of an application failure, or if for any reason the application goes down, the target DN, label, and unicode label will reset to the default. JTAPI provides the interface resetIntercomTarget () on the CiscoIntercomAddress to reset the intercom target.

Auto-answer always stays enabled for CiscoIntercomAddress. The application can invoke the method getAutoAnswerEnabled () on CiscoAddress to get the auto-answer capability of an address.

For an intercom target that is connected with one-way media to the Intercom initiator, the device state would be set to CiscoTermDeviceStateWhisper. This is a new device state for the terminal object. In this state, the terminal can initiate a new call or receive a new incoming call. If the application enables a filter to receive this device state, the application receives CiscoTermDeviceStateWhisperEv. The application can enable a filter by calling setDeviceStateWhisperEvFilter() on CiscoTermEvFilter. The DeviceStates DEVICESTATE_ACTIVE, DEVICESTATE_HELD, and DEVICESTATE_ALERTING all override DEVICESTATE_WHISPER; if one call exists in active, held, or alerting state, and another in whisper, the DeviceState will be DEVICESTATE_ACTIVE, DEVICESTATE_HELD, or DEVICESTATE_ALERTING, respectively.

Note

The Cisco Unified JTAPI implements the javax.telephony.TerminalConnection interface join() to let the intercom target talk back to the initiator. The system implements this interface for CiscoIntercomAddresses only. If applications invoke this interface for regular shared lines in a passive or bridged state, JTAPI throws a MethodNotImplimented exception.

Tip	This feature is backward compatible if the application-controlled devices (terminals) do not have intercom lines configured on them. Applications can disable the intercom feature by not having an intercom line configured on the application-controlled devices (terminals).

For detailed information about these interface changes, see the following topics:

• CiscoHuntConnection

• CiscoAddrIntercomInfoRestorationFailedEv

• Related Documentation

• CiscoCall

• CiscoProvider

• CiscoTermEvFilter

• CiscoTerminal

• CiscoTerminalConnection

• CiscoTermDeviceStateWhisperEv

In Release 6.0(1) of Cisco Unified Communication Manager, support for intercom feature was added. Intercom feature requires destination to be auto-answered with one-way audio; therefore, no shared addresses can be configured for intercom. When user logs in by using Extension Mobility (EM) profile, it is possible to end up with shared address for intercom; so, currently extension mobility is not supported with intercom. Due to the wide use of extension mobility, this CIA is addressing the need to support intercom for extension mobility while still maintaining the single destination nonsharable nature of intercom addresses.

This feature requires intercom addresses to be configured with default terminal, and allows configuring of intercom address on EM profile. When EM user logs in to a terminal with EM profile that is configured with an intercom address, intercom address is available only if default terminal of intercom address is same as terminal where user has logged in. If an intercom address is configured on terminal but default terminal for intercom address is not that terminal, intercom address does not appear on terminal. If this terminal is configured in the control list of Cisco Unified JTAPI application, JTAPI does not create intercom address in the provider domain. From Cisco Unified JTAPI point of view, there is no new interface or changes to support this feature. However, this feature introduces some transitional scenarios where intercom functionality may not work on intercom addresses. See the use cases.

This feature provides support for IPv6 addresses and CiscoUnified JTAPI is enhanced to support IPv6 connectivity to CTIManager. It enables Cisco Unified JTAPI applications to see the IPv6 address as the calling party address if the IPv6 support feature is enabled and if the Calling Party is using an IPv6-enabled phone. This feature support the following functions:

• Cisco Unified JTAPI exposes new API canSupportIPv6() on the CiscoProviderCapabilities Interface to indicate whether Cisco Unified Communications Manager configuration is supporting IPv6.

• Cisco Unified JTAPI closes the media or route terminal if there is mismatch between what has been previously registered and what is currently configured. CiscoTermRegistrationFailedEv and the new reason code IP_ADDRESSING_MODE_MISMATCH are then sent as per this scenario.

• The IP Address capability of the Terminal is exposed by API getIPAddressingMode() on the CiscoTerminal Interface. The IP Address capability is available on CiscoTerminal/CiscoMediaTerminal and CiscoRouteTerminal.

• The IPv6 calling party IP address is provided through the Cisco extensions of CallCtlConnOfferedEv and RouteEvent in an InetAddress object as well as the IPv4 address for IPv4-enabled devices.

The RTP Address in CiscoRTPOutputStartedEv and CiscoRTPInputStartedEv also has an IPv6 address in case the observed device is an IPv6 device. That is, the API getLocalAddress() on CiscoRTPInputProperties and the API getRemoteAddress() on CiscoRTPOutputProperties can now return an IPv6 format IP Address. The API returns an InetAddress object, and applications can verify that it is an instance of Inet4Address or Inet6Address to determine if it is an IPv4 or IPv6 format IP Address.

Applications must reset the devices after their IP Addressing Mode is changed, otherwise there might be ambiguity in the expected results.

From Release 7.1, Cisco Unified JTAPI provides getIPAddressingMode() API on CiscoTerminal. The getIPAddressingMode() API for CTI Ports and Route Points are also supported from this release.

Cisco Unified JTAPI extends the same API on CiscoTerminal and it returns the configured IP addressing mode of the IP phone on the Cisco Unified Communications Manager Admin pages. If the user modifies the IP Addressing mode from the Cisco Unified Communications Manager Admin pages after the device is registered, the device must be reset. The updated value from Cisco Unified JTAPI is exposed only after the IP phone is reset. If the configured IP Addressing mode supports both IPv4 and IPv6 addresses, the phone may be registered with either of these addresses or with both. This depends on conditions such as network type and Cisco Unified Communications Manager support for IPv6. So, if the IP Addressing mode mode supports both IPv4 and IPv6 addresses, getIPAddressingMode() on CiscoTerminal returns CiscoTerminal.IP_ADDRESSING_MODE_IPV4_V6.

This enhancement provides support for iSac codec and enables the application to register CiscoMediaTerminal or CiscoRouteTerminal with iSac codec capability. For this codec, frame size and bit rate are variable and determined dynamically. Applications do not set these values.

The bit rate and packetSize that are exposed on interface CiscoRTPInputProperties and CiscoRTPOutputProperties will not be a constant for this codec, so application logic should not rely on these values if codec (payloadType) is iSac.

The Java Socket Connect Timeout enhancement enables the configuration of a timeout in seconds by using the Cisco Unified JTAPI specification and prevents connection delays to the CTIManager when the primary CTI Manager. The default is 15 seconds.

If the default of 15 seconds is unacceptable to the application, the default JAVA API of zero (0) sets the behavior to the normal JAVA Socket Connect API.

The values range from 5 through 180 seconds. Zero defaults to Java behavior of the socket connect without any time-out for connection.

In this version, this feature allows applications to conference two calls that are on different addresses of the same terminal. It will also let applications add participants to a conference using a noncontroller. Join across lines is not supported on CTI-supported phones that run SIP.

You can disable join across lines feature by turning off Join Across Lines Policy service parameter, while you can disable Conference Chaining and feature to allow noncontroller adding participant to conference by disabling the "Advanced Ad Hoc Conference Enabled" and "Non-linear Ad Hoc Conference Linking Enabled" service parameters.

Note	Join Across Lines is supported only on phones that run SCCP.

The Join Across Lines feature allows support for conference across lines. It allows two or more calls on different addresses of the same terminal to be joined though the join softkey on the phone or conference() API that JTAPI provides. The behavior to JTAPI applications change, as applications do not perceive a common controller in final and consult calls.

There is no change in the API and the same events are delivered whether calls are conferenced on the same address (regular conference) or across addresses (Join across lines). When join across lines feature is performed CiscoConferenceStartEv/EndEv will be provided to all addresses on the controller terminal that have consult or final calls that are being joined together into one conference.

In CiscoConferenceStartEv, the conferenceControllerAddress will always be the primary controller address. Application can now set the controller via the setConferenceController() API. If application does not specify this, then JTAPI itself would find a suitable controller for the conference. Cisco recommends that applications set the controller address when Join Across Lines feature is invoked.

If observer is not added on the controller address, applications may see null values for either the talking or held terminal connection values in the CiscoConferenceStartEv. Before this release, when application tried a conference across lines, the request failed at the JTAPI layer itself. With this release, the conference() API implementation enhances all requests to pass through after finding suitable terminal connections of the final and consult calls. JTAPI relies on the common terminal of the addresses involved in the call to find suitable terminal connections. Multiple conference across address is also supported when more than two calls need to be joined. SIP devices in 5.1.2 release do not support this feature. JTAPI throws exception (ILLEGAL_HANDLE) if this feature is requested on a SIP device.

There are no interface changes for this feature. Behavior changes with respect to events provided to applications.

Join Across Lines feature supports on CTI-supported SIP phones and SCCP phones. The enhancements are:

• Applications can conference two calls in which each conference is on a different address but on the same terminal.

• Add participants to a conference using a non-controller.

Note

You can disable Join Across Lines by turning off the Join Across Lines Policy service parameter. Conference Chaining and the feature that allows Non-Controller adding participant to conference can be disabled by disabling the Advanced Ad Hoc Conference Enabled and Non-linear Ad Hoc Conference Linking Enabled service parameters.

The following behavior occurs when an application issues a conference request, but selected and active calls are not part of the conference request. It also applies for user-selected calls that are not part of the conference request, but become part of the resulting conference:

• The Active Call on a Terminal is always added to the resulting conference when conference is invoked on a call on any address on that terminal. Consider that B1 and B2 addresses exist on the same terminal, then:

  • A --> B1- GC1

  • C --> B1- GC2

  • D --> B2- GC3 (active call)

  The application invokes GC1.conference (GC2) and results in A-B1-C-D in a conference with GC1, although the call with D was not part of the conference request.

  An active conference call on a terminal is added to the resulting conference when conference is invoked on a call on any line on that terminal. In this case, the active conference call becomes the surviving final call (provided the application-specified primary call is not a conference call).

  In this example, the application specified primary call is cleared after the conference operation. It is possible that the application-specified primary call may not join the resulting conference and in that case the call is not cleared after the conference is complete.

• Consider that the B1 and B2 addresses on the same terminal and conf1 is a conference call with A-B1-C in conference with B1 as the controller, then:

  • B1 --> D – GC1 (on hold)

  • conf1 – GC2 (active call)

  • B2 --> E – GC3 (on hold)

  Application invokes GC1.conference(GC2, GC3). This results in A-B1-C-D-E in conference with GC2 as the surviving call. Although application had specified GC1 to be the primary call, GC1 does not survive after the conference.

The behavior also applies to regular conferencing with a common controller. Consider A, B, C, and D are lines on different terminals, then:

• A --> B - GC1

• C --> - GC2

• D --> - GC3 (active call)

The application requests GC1.conference (GC2). This results in A-B-C-D in conference with GC1. Although a direct call with Dwas not part of the conference request, D joins the conference.

This release of the CiscoJTAPIClient supports only JRE 1.4. There are no interface changes; however, the JRE 1.2 and 1.3 versions are no longer supported. This change is to support QoS, which is available only in the JDK1.4 version (and above). Inaddition, jtapi.jar contains Cisco encryption files that depend on the JRE 1.3 version (and above). This provides a stronger password encryption algorithm when it is sent over TCP to CTIManager. As part of this feature, JTAPI invokes the API provided by IMS (Identity Management System, a Cisco Unified Communications Manager component) to encrypt a password before sending it.

JRE 1.4 also enables Cisco Unified JTAPI to use additional JDK 1.4 APIs. Applications that use previous versions of JRE must install JDK 1.4 to use Cisco Unified JTAPI.

Note

There are no interface changes to JTAPI Applications, however JTAPI.jar contains RSA jsafe.jar (3.3) and Apache log4j-1.2.8.jar files. If Applications are using any jar files that are not compatible with these versions of jsafe.jar (Version 3.3) and log4j-1.2.8.jar, then JTAPI or the Application may not work, depending on which one is in the classpath first

As part of this migration, JTAPIPreferences and sample applications dependency on MS-JVM was also removed. Two new configuration parameters were provided on the Advanced tab in the JTAPI Preferences dialog box:

• JTAPI Post Condition Timeout

• Use Progress As Disconnected

  Backward compatibility

This feature is not backward compatible.

In order to connect to Release 5.0 of Cisco Unified Communications Manager Administration, JTAPI clients have to upgrade to the new version of JTAPI bundled with the Cisco Unified Communications Manager Administration Release 5.0. JTAPI version is in the form of 3.0(X.Y), where X and Y depend on the sub-release. Applications cannot connect with prior release of JTAPI.

This feature removes currently supported languages for Cisco Unified JTAPI client install. Cisco Unified JTAPI client install is only supported in English. It also adds the capability to dynamically update the locale in JTAPI Preference application from the Cisco Unified Communications Manager server. JTAPI Preference application will continue to support all the languages that are supported in prior releases. Support for adding new languages and updating locale files is also added.

Before this release, the Cisco Unified JTAPI client install and JTAPI Preferences application were localized during builds and did not add support for new languages or update locales for existing languages. The JTAPI client locale updates were performed in Cisco Unified Communications Manager maintenance releases. This feature adds capability to dynamically update locale file for JTAPI Preferences application, and JTAPI Client install is installable only in English languages.

The JTAPI Client install needs the Cisco Unified Communications Manager TFTP server IPaddress. The TFTP IP address is used for downloading locale files for the preferences application. If the TFTP IP address is not entered or an incorrect IP address is entered, the preference application displays only in English language. Further on, whenever new locale updates are available, JTAPI Preferences application will notify user about available updates and update locale files.

This feature enables administrators to configure geographic locations and restrict calls that pass through a PSTN gateway to be connected directly to a VoIP phone or VoIP PSTN gateway in another geographic location. This feature allows use of single line analog phones and remains compliant with the Telecom Regulatory Authority of India (TRAI) regulation.

This feature can be turned off by using the Logical Partitioning Enabled service parameter, which is disabled by default.

This feature enables multiple active calls at the route point, and applications can terminate media for all active calls by specifying the IP address and port number for each call or whenever media is established.

To use this feature, applications must register the route point by supplying media capabilities. When a call gets answered at this route point, CiscoMediaOpenLogicalChannelEv gets sent to the applications. This event gets sent whenever media is established. Applications must react to this event and specify the IP address and port number where they want to terminate media.

A CiscoRouteTerminal represents a special kind of CiscoTerminal that allows applications to terminate RTP media streams. Unlike a CiscoTerminal, a CiscoRouteTerminal does not represent a physical telephony endpoint, which is observable and controllable in a third-party manner. Instead, a CiscoRouteTerminal represents a logical telephony endpoint, which may get associated with any application that intends to route calls and also terminate media. Unlike CiscoMediaTerminal, CiscoRouteTerminal can have multiple active calls at the same time. Typically, CiscoRouteTerminals get used to place calls in queue until an agent is available to service the caller.

Note	Only RoutePoint Terminals appear as CiscoRouteTerminal through JTAPI.

Terminating media comprises a three-step process.

1. The application registers its media capabilities with this terminal by using the CiscoRouteTerminal.register method.

2. An application adds an observer that implements CiscoTerminalObserver interface by using the Terminal.addObserver method.

3. The application must add addCallObserver on CiscoRouteTerminal or on CiscoRouteAddress to receive CiscoCall object from the provider by using CiscoRTPHandle.

Applications receive CiscoMediaOpenLogicalChannelEv for each call and must supply the IP address and port number by using the setRTPParams method on CiscoRouteTerminal.

You must modify applications that are written for the CiscoJtapiClient 1.4(x) release or earlier to register with CiscoRouteTerminal. NO_MEDIA_TERMINATION if the applications are not interested in media termination.

Multiple applications can register with the same route point as long as they are registered with the same media capabilities and registrationType. All applications, if they have registered with CiscoRouteTerminal.DYNAMIC_MEDIA_REGISTRATION and then add a terminal observer, receive CiscoMediaOpenLogicalChannelEv, but only one application can invoke setRTPParams.

Note	Applications that terminate media must use the CallControl package for answering and redirecting calls. Applications that only route calls can use a routing package.

Note

Applications should be aware that, if any features are performed before reacting to CiscoMediaOpenLogicalChannelEv, the features may fail. If applications do not respond to these events in the time that is specified in the Media Exchange Timeout parameter in the Cisco Unified Communications Manager Administration windows, the call may fail.

The following new or changed interfaces exists for Media Termination at Route Point:

For details on these interfaces, see Cisco Unified JTAPI Extensions To view the message flow for media termination at route point, see Message Sequence Charts

The media termination feature allows applications to transmit and capture the bearer of a call, for example, audio or video. This action sometimes gets referred to as "rendering and recording" or "sourcing and sinking" media. It remains distinct from call control because media termination concerns the data that flows between endpoints in a call, not the details of setting up or tearing down calls. For example, an automatic call distributor (ACD) uses call control to route calls among available agents but does not terminate media. An interactive voice response (IVR) application, on the other hand, uses call control to answer and disconnect calls and uses media termination to play sound files to callers.

Although no telephony applications are solely interested in media termination, this feature always gets used in combination with call control. JTAPI 1.2 primarily represents a call control specification and offers very limited support for applications that require media termination. Because the Cisco Unified Communications Solutions platform supports media termination to a much greater degree than JTAPI standard, the Cisco Unified JTAPI implementation extends JTAPI to add full support for this feature.

In Cisco Unified JTAPI, software-based media termination occurs by using Computer Telephony Integration (CTI) ports. They include one or more lines (dialable numbers) that can be used to originate or receive calls. They however need a controlling application to provide the source and sink of the media. An application registers its interest in the media termination port with the Cisco Unified Communications Manager. The Cisco Unified Communications Manager then delivers all the events that relate this virtual device to the application. InCisco Unified JTAPI, CTI ports get referred to as CiscoMediaTerminals. The following figure shows the CTI port configuration. For details about administering and configuring a CTI port, refer to the Cisco Unified Communications Manager Administration information.

To implement a softphone application (where the PC acts as the telephone set, for example), the Cisco Unified JTAPI application would manage a CTI port.

The Enhanced Message Waiting Indicator (MWI) feature enables applications to provide the following message counts to be displayed on phones that support the enhanced message waiting counts:

• Total number of new voice messages (includes normal and high priority messages)

• Total number of old voice messages (includes normal and high priority messages)

• Number of new high priority voice messages

• Number of old high priority voice messages

• Total number of new fax messages (includes normal and high priority messages)

• Total number of old fax messages (includes normal and high priority messages)

• Number of new high priority fax messages

• Number of old high priority fax messages

Two new APIs are added as CiscoAddress JTAPI extensions to provide the enhanced MWI message summary information. Similar to the existing setMessageWaiting APIs, one of the APIs allows summary information to be set up for the observed address. The other API allows message summary information to be set up on any address that is reachable on the observed address, as defined by the configured calling search space of the observed address.

These new APIs can also be used on phone types that do not support the enhanced message counts. If used on non-supported phones, these APIs behave similar to the existing setMessageWaiting method, that is, only the messaging waiting indicator lamp is turned on or off and counts are not displayed.

This feature enables applications to modify the calling party DN in the select route API from the route point. Applications may pass an array of modifying calling numbers in the selectRoute API and an array length of modifying calling numbers may equal the length of the route that is selected. If no modifying calling number element is present for a corresponding routeSelected index or if the element is null, then no modifying calling number gets set for that route selected element.

Two new interfaces getModifiedCallingAddress () and getModifiedCalledAddress () are exposed on the call object, which returns modified calling or called number. If no modification occurs, these interfaces may return the same values as getCurrentCallingAddress () and getCurrentCalledAddress () interfaces. If an application is only controlling the route point and modifies the calling number by using selectRoute API, it may not get modified calling address in the getModifiedCallingAddress interface. If an application is controlling any calling or called parties, it may get correct values after it receives call control events after the calling number is modified.

A new interface, getRouteSelectedIndex (), gets exposed on the new class CiscoRouteUsedEvent, an extension of RouteUsedEvent, which gives the index of the selected route. Applications need to cast the RouteUsedEvent to the CiscoRouteUsedEvent to get access to this method.

routeSelected[0]  =  133555 
routeSelected[1]  =  144911 
routeSelected[2]  =  143911 
routeSelected[3]  =  5005  

modifiedCallingNumber[0]  = null
modifiedCallingNumber[1]  = 9721234567
modifiedCallingNumber[2]  = 9721234568
modifiedCallingNumber[3]  = null

If routeSelected[0] or routeSelected[3] is selected for routing, the modifying calling number may not get applied.

You can only use this feature after an administrator enables the modifying calling number check box in the Cisco Unified Communications Manager Administration for a particular user, which by default is False. If it is not configured, a RerouteEvent with the cause of RouteSession.CAUSE_PARAMETER_NOT_SUPPORTED gets sent to the applications. The application that is modifying the calling number needs to be aware that display name on the called party is affected, and subsequent feature interactions of the calling or called party may result in inconsistent behavior.

The following new or changed interfaces exist for Modifying Calling Number:

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for Modifying Calling Number, see Message Sequence Charts.

Cisco Unified Communications Manager enables the use of supplementary services by phones that are configured for Multilevel Precedence and Preemption (MLPP). Cisco Unified Communications Manager does this by maintaining the precedence level for calls.

Note	JTAPI does not provide the precedence level of applications.

Multiple calls per DN represent the ability to support multiple calls on a line (DN) and the features operation on these calls. Prior to Cisco Unified Communications ManagerRelease4.0(1), the system supported a maximum of only two calls. Cisco JTAPI now supports multiple calls per line, which allows multiple calls on the same line and feature operation on that line.

No interface or message flow changes occurred for Multiple Calls Per DN.

It is very common in a Cisco Unified CM deployment that a hunt pilot has more calls distributed through the call distribution feature than its hunt members can handle at any given time. Native Queuing feature holds the calls in a queue until they are answered. When a hunt member is available, the call is removed from the queue and offerred to the hunt member.

To enable this feature, the Cisco Unified CM administrator needs to enable the check box Queue Calls in Queuing section of the Hunt Pilot configuration page. Following settings are available under Native Queuing feature configuration:

• Maximum Number of Callers Allowed in Queue (1–100): This is the queue depth configuration and reflects the maximum number calls that can be in the queue at any point of time.

  • Destination When Queue is Full: User Configurable Destination number to which the calls are forwarded when the Maximum Number of callers allowed in queue limit is reached.

  • Disconnect: This option results in the call getting rejected and dropped when the Maximum Number of callers allowed in queue limit is reached.

• Maximum Wait Time in Queue (10–3600 seconds): User configurable Maximum wait time a call can be in the queue.

  • Destination When Maximum Wait Time is met: User Configurable destination DN to which the call is forwarded when the maximum wait time in queue is reached.

  • Disconnect: This option results in the call getting rejected and dropped when the the maximum wait time in queue is reached.

• When There Are No Hunt Members Logged In or Registered: User configurable destination DN to which the queue feature forwards the calls when none of the hunt members in the HuntPilot are registered or logged in.

  • Disconnect: This option results in the call getting rejected and dropped when there are no hunt members available for the dialed hunt pilot.

  • Destination: User Configurable destination DN to which the call is forwarded when no hunt members available for the dialed hunt pilot.

If a caller calls a hunt pilot with all its members busy, a CiscoHuntConnection will be created temporarily. Then, when this feature is enabled, the hunt connection will drop and a new connection will be created which will have the address name same as that of the hunt pilot and the address type will be CiscoAddress.INTERNAL. This new connection will be moved to CallControlConnection.QUEUED state and it will remain in this state until the call gets dequeued or dropped.

Cisco JTAPI exposes the following new reasons:

• CiscoFeatureReason.REASON_QUEUING

• CiscoFeatureReason.REASON_DEQUEUING

• CiscoFeatureReason.REASON_DEQUEUING_TIMER_EXPIRED

• CiscoFeatureReason.REASON_DEQUEUING_AGENTS_BUSY

• CiscoFeatureReason.REASON_DEQUEUING_AGENTS_UNAVAILABLE

The above reasons indicate when a call gets enqueued and dequeued respectively because of the various configurations on the Hunt Pilot.

Note

The behavior is different when conferencing a queued calls. If a caller which is in a queue conferences the call with another party, then the queued connection is dropped and a new connection is created with the address of the hunt pilot number and the address type is CiscoAddress.UNKNOWN, and it will be moved to CallControlConnection.ESTABLISHED state. If a call is removed from the queue, for example when an agent becomes free, the agent will be added to the conference and a connection will be created for it. For the Hunt Pilot, JTAPI creates a normal connection instead of a CiscoHuntConnection. This is a limitation of JTAPI in handling a conference with Hunt Pilot. In a case where only the Hunt member is observed, there will be no issues and JTAPI will be able to handle it.

In earlier releases of CiscoJTAPI (CiscoJTAPI versions 1.4(x.y)), when a call was made to an address outside of the cluster, CallCtlConnNetworkReachedEv and CallCtlConnNetworkAlertingEv events were delivered to the farend address.

In later versions of Cisco Unified Communications Manager (4.0 and above) and Cisco Unified JTAPI (2.0), these events were not delivered. In these versions CallCtlConnection for the farend address went to the ESTABLISHED state from the OFFERED state. The previous versions of Cisco Unified JTAPI delivered CallCtlConnOfferedEv, CallCtlConnEstablishedEv for the farend address when a call was made across a gateway with "overlap sending" turned off. CallCtlConnNetworkReachedEv and CallCtlConnNetworkAlertingEv events were not delivered to the application.

In Cisco Unified Communications Manager4.0 and 4.1, the "Allow overlap sending" flag on the route pattern configured for the gateway or the "AllowNetworkEventsAfterOffered" parameter in jtapi.ini needed to be turned on to receive network events.

In Cisco Unified Communications ManagerRelease 5.0, if the "Allow overlap sending" flag is enabled, an application sees ConnCreatedEv, CallCtlConnNetworkReachedEv, CallCtlConnNetworkAlertingEv, and CallCtlConnEstablishedEv for the farend address for calls across a gateway.

If the "Allow overlap sending" flag is not enabled, an application sees ConnCreatedEv, CallCtlConnOfferedEv, CallCtlConnNetworkReachedEv, CallCtlConnNetworkAlertingEv, and CallCtlConnEstablishedEv for the farend address for calls across a gateway.

Note	AllowNetworkEventsAfterOffered is not available in Cisco Unified Communications Manager Release5.0. The above events are delivered regardless of the jtapi.ini parameter setting.

This event is sent to application when TerminalRegistration fails for some reason. The return value of getErrorCode() interface indicates the type of failure. On receiving this event, application should try to reregister the Terminal. In this version a new return value is added to this interface. CiscoTermRegistraionFailedEv.UNKNOWN is introduced in this version to handle unknown failures.

Any party in a conference can now add participants into the conference. In previous releases, only the conference controller could add participants.

• CiscoConferenceStartEv contains an identifier for the requestor party.

• The method getConferenceControllerAddress returns the terminal connection of the requestor.

• The new method getOriginalConferenceControllerAddress() for CiscoConferenceStartEv returns the terminal connection of the original controller.

Cisco Unified JTAPI applications can register to receive events when calls are parked and unparked. CiscoProvCallParkEv events will be delivered to provider observer when the application registers for this feature. To successfully register for this feature, ensure that the "call park retrieval allowed" flag for the user is turned on. You can access this flag with the user configuration on Cisco Unified Communications Manager Administration. After registering for this feature, the application will receive CiscoProvCallParkEv events whenever a call is parked or unparked from any device in the cluster.

The following new interfaces allow applications to register and unregister for this feature:

public interface CiscoProvider {
public void registerFeature ( int featureID ) throws 
     InvalidStateException,  PrivilegeViolationException;
public void unregisterFeature ( int featureID ) throws 
     InvalidStateException;
}

The featureID is CiscoProvFeatureID.MONITOR_CALLPARK_DN.

This feature provides a new park reversion behavior to applications invoking park request. Currently, when the park reversion timer expires, the call is reverted to the address of the parker. With the new behavior, the call remains parked at the park DN, even as the Park Monitoring reversion timer expires.

This feature also enables status monitoring of the parked call at the address of the parker. After a call is parked using the existing CiscoConnection.park() JTAPI API on newer phones or directly from the phone itself, Cisco Unified JTAPI delivers a new event CiscoAddrParkStatusEv, which includes the current status of the parked call. The application must then add AddressObserver on the address of the parker, and enable a filter to receive this event. If application adds an observer after the call is parked, then the events are delivered with CAUSE_SNAPSHOT. The park status in the new event can be one of the following:

• Parked—Indicates a call was parked by the user of the application.

• Reminder—Indicates the park monitoring reversion timer for the parked call has expired.

• Retrieved—Indicates a previously parked call was retrieved.

• Abandoned—Indicates a previously parked call is disconnected while waiting to be retrieved.

• Forwarded: indicates the parked call has been forwarded to the configured Park Monitoring Forwarded No Retrieve destination, as the Park Monitoring Forward-No-retrieve timer has expired.

When the cause is CAUSE_SNAPSHOT the park status can be either Parked or Reminder state only.

On the phone, these notifications are targeted, that is, only the device parking the call can see these notifications (devices sharing line with the parker's device does not receive similar notifications). In Cisco Unified JTAPI, getTerminal() interface on CiscoAddrParkStatusEv has been added tomanage this. This returns the terminal on whose address, these notifications were received and this is the terminal that parked the call.

Cisco Unified JTAPI also provides the CiscoCallID to applications in this new event. Applications may use this to retrieve the call object. However CiscoCallID.getCall() may return null value if the call does not exist in the provider's domain at the time this event is received.

Cisco Unified JTAPI provides a new interface CiscoAddrEvFilter to control or filter the new event notifications to applications. Applications may get or set the filter value through the APIs getCiscoAddrParkStatusEvFilter() and setCiscoAddrParkStatusEvFilter() on the CiscoAddrEvFilter interface. Two new methods, getFilter() and setFilter(), have also been provided in the CiscoAddress to get and set the values of the filters in the CiscoAddrEvFilter interface. Applications receive the new event notification CiscoAddrParkStatusEv only if the filter is enabled and the setFilter() is invoked on CiscoAddress. By default, the filter value for CiscoAddrParkStatusEvFilter is false to maintain backward compatibility.

When a call is parked, the Park monitoring reversion timer starts and then expires. After this, Park Monitoring Forward No Retrieve timer starts. When this timer expires, and the Forward No Retrieve destination is configured, the call is forwarded to this destination. A new CiscoFeatureReason FORWARD_NO_RETRIEVE is delivered in the connection events, when connections are created at the forwarded destination. If the Forward No Retrieve destination is not configured, call is forwarded back to the parker's DN, with the same reason as when park reversion occurs (CiscoFeatureReason.PARKREMINDER).

When application invokes CiscoAddress.getAddressCallInfo(Terminal term), the CiscAddressCallInfo which is returned is now enhanced to include number of parked calls. This returns the number of parked calls. Cisco Unified IP Phone 7900 Series with SIP/SCCP returns zero value even if there are calls parked by this address.

This feature is applicable only when newer phones park the call. If Cisco Unified IP Phone 7900 Series with SIP/ SCCP, parks the call, user continues to see the existing behavior. So, if a Cisco Unified IP Phone parks the call and is sharing a line with a Cisco Unified IP Phone 7900 Series with SIP, the new Park Monitoring enhancements can be seen. However, if the Cisco Unified IP Phone 7900 Series with SIP or SCCP invoked park, the old Park behavior would be seen on all the phones, if application is monitoring any of these lines.

Users can set the Park Monitoring Reversion Timer to zero and set the Park Monitoring Forward No Retrieve Destination to the existing Park Reversion Duration timer to get the old behavior on the Cisco Unified IP Phone (provided the Forward No Retrieve destination is not configured) if the user so desires. However, the event notification cannot be controlled.

On Cisco Unified Communications Manager Service Parameter pages, the timers mentioned above can be configured. These would apply only for SIP versions of future models of Cisco Unified IP Phone .

Park Monitoring Reversion timer: This timer is started as soon as the call is parked. This is the amount of time that a call remains parked before the user is reminded that there is a parked call. The range is 0-1200 seconds, with default value of 60 seconds.

Park Monitoring Periodic reversion timer: The frequency in which the user is reminded about the parked call. The range is 0-1200 seconds, with default value of 30 seconds.

Park Monitoring Forward No Retrieve timer: This timer is started when the park monitoring reversion timer expires. This is how long, in seconds, the park reminder notification plays before the parkee is redirected to the parker's Park Monitoring Forward No Retrieve (FNR) destination. The range is 30-1200 seconds, with default value of 300 seconds.

Park Monitoring Forward No Retrieve Destination is configurable on the line page in Cisco Unified Communications Manager Line page settings.

Assisted DPark provides an alternative one step way to perform DPark operation on phones. When user performs Assisted DPark from newer phones and application is monitoring the parked party, Cisco Unified JTAPI provides reason CiscoFeatureReason.REASON_REFER in the connection events (ConnCreatedEv, ConnInProgressEv and CallCtlConnQueuedEv) for DPark DN. Currently when DPark is done, application gets connection events with CiscoFeatureReason.REASON_TRANSFER.

When a parked call is not retrieved for a specified time, a reminder call returns to the address that parked the call, and Park Number connection moves to the Disconnected state. The call reconnects and moves to the Established state. A terminal connection in Talking state gets created for the address that parked the call.

When a call is parked from an IP phone, the park number displays on the phone. Any terminal can unpark the call by dialing the park number. When a call is unparked, a new call gets created with connections to unparked address. The CallControlConnection for the park number in the original call, which is in the Queued state, moves to the Disconnected state.

Prior to Cisco Unified Communications Manager Release 5.0, JTAPI did not support partitions. JTAPI considered addresses with the same DN, but different partitions, as same address. It created only one Address object for such cases because addresses are identified only by their DN and not by their partition information.

Beginning with Release 5.0, JTAPI supports addresses that have the same DN but belong to different partitions and treats them as different addresses. Partition information of the addresses is exposed to applications through the methods specified below. Applications that want to make use of this partition support feature must use the API provided to them through JTAPI interfaces and use the address objects accordingly.

This feature is backward compatible. JTAPI supports the current APIs that are used to open and access address objects.

In Cisco Unified Communications Manager Release 5.0, JTAPI is partition aware, and the following configurations are supported.

• Addresses with the same DN, in the same partition, and in different devices get treated as shared lines.

• The system does not allow addresses with the same DN, in the same partition and in the same device.

• Addresses with the same DN, in different partitions, and in the same device get treated as different addresses. Two address objects get created for this scenario, and the application can distinguish between the two by calling the getPartition() API on the address objects.

• Addresses with the same DN, in different partitions, and in different devices get treated as different addresses. Two address objects get created for this scenario and the application can distinguish between the two by calling the getPartition() API on the address objects.

Partition support changes in JTAPI are confined to the address objects and do not affect any other functions or classes of JTAPI. The following sections specify the interface changes.

CiscoAddress Interface

A new method is provided in this class with the following signature.

string getPartition ()

Returns the partition string of the address object. Applications need to use this method to get the partition information. JTAPI uses this partition information to distinguish between addresses that have the same DN but belong to different partitions and sends the partition information to open the specific addresses.

For example, a provider open returns two addresses, A(1000, P1) and B (1000, P2), where A and B denote the address objects, 1000 denotes the DN of the address objects, and P1, P2 indicate the partitions to which the addresses belong.

When the user invokes A.getPartition (), P1 gets returned while B.getPartition () returns P2.

The provider.getAddresses() method returns multiple addresses in which the Address objects have the same DN but different partition information. An Application can use this method to distinguish between two Address objects that have the same DN but belong to different partitions.

CiscoProvider Interface

The CiscoProvider interface provides the following methods:

Address[]	getAddress(String number) Returns an array of Address objects that corresponds to the number and different partitions.
Address	getAddress(String number, String partition) Returns the Address object that has the same DN as the number parameter and belongs to the same partition as specified by the partition parameter.

If two addresses A(1000, P1) and B(1000, P2) exist, where A and B denote the address objects, 1000 denotes the DN of the address objects, and P1, P2 indicate the partitions to which the addresses belong, when an application calls provider.getAddress("1000"), it gets two address objects, A and B.

When the application calls A.getPartition(), it returns P1, B.getPartition() returns P2, and so on. An Application can distinguish between the two address objects that are using the getPartition method.

Consider the case where the application calls provider.getAddress(1000, P1). In this case, the application specifically looks for the address object whose DN is 1000 and partition is P1. In this case, "A" gets returned by the provider object.

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message sequences for partitions support, see Message Sequence Charts

The administrator can use the CUCM Admin Panel to configure options for login credentials. The password expiry configuration allows the administrator to specify the following two parameters:

1. The time before the password expires (in days) and

2. The number of days before the end of the password expiry to alert the user to change the password.

If a password is expired, JTAPI delivers an exception to the application. In a situation where a password is going to expire soon, JTAPI delivers a new event to the application. JTAPI does not allow applications to modify any of these values, it only reports the information.

Persistent Connection is an extension Cisco Extend and Connect feature that was implemented in Unified Communications Manager Release 9.1. A persistent call refers to a call between the Unified Communications manager (CTI Remote Device) and a remote destination that stays up even after calls to it are dropped. JTAPI APIs and error codes were added.

JTAPI supports a new API, CiscoAddress.createPersistentCall(), which allows applications to create persistent calls. At least one remote destination must be configured and the active remote destination must be set. There can be only one persistent call per remote device. Persistent calls cannot be created if there is already a call on the remote device; otherwise, the application receives CiscoJtapiException.OPERATION_NOT_AVAILABLE_IN_CURRENT_STATE. Furthermore, no feature invocations are allowed on or involving persistent calls (park, hold, conference, and transfer).

Two new JTAPI APIs return information about the persistent call. The CiscoAddress.getPersistentConnection() API returns the connection object that is associated to the persistent call. It returns null if no persistent call exists. This API also allows you to check if an address has a persistent connection created on it and from there you can get the call object. The other newly added API is CiscoCallisPersistentCall(), which returns true if the call is a persistent call and false if the call is a normal call.

Existing JTAPI APIs such as Provider.getCalls(), Address.getConnections(), and Terminal.getTerminalConnections() return only the information for normal calls and do not return anything for the persistent call. Provider.getCalls() returns all the calls that are associated with the provider, excluding the persistent calls. Address.getConnections() returns all the connection objects that are associated with this address, excluding the connection for the persistent call. Terminal.getTerminalConnections() returns all the terminal connection objects that are associated with this device, excluding the terminal connection for the persistent call. This functionality helps with backward compatibility so applications do not need to make any changes to their current implementations.

No new APIs are added to disconnect the persistent calls. Existing Call.drop() and Connection.disconnect() JTAPI APIs can be used to disconnect or drop the persistent calls. Persistent calls cannot be dropped if there is an active call to the remote device. Persistent calls can also be dropped in any of the following scenarios:

• The call is dropped by the remote destination (the remote destination hangs up).

• The remote destination is no longer active. If there is an active call, as soon as that call is over, the persistent call will drop.

After they are created, persistent calls remain connected until the maximum call duration timer expires in which case the call will be cleared.

Some of the new JTAPI Error Codes introduced as part of this feature include the following:

• CiscoJtapiException.CTIERR_CREATE_PERSISTENT_CALL_FAILED: Indicates that there is an issue with creating a persistent call.
• CiscoJtapiException.CTIERR_PERSISTENT_CALL_EXISTS: Indicates that a persistent call already exists.
• CiscoJtapiException.CTIERR_OPERATION_NOT_ALLOWED_ON_PERSISTENT_CALL: Indicates that the specified operation is not allowed on a persistent call.
• CiscoJtapiException.CTIERR_DISCONNECT_PERSISTENT_CALL_FAILED_CALL_ACTIVE: Indicates that the request to disconnect the persistent call failed because there is an active customer call. Only when there are no active calls can the persistent call be disconnected.
• CiscoJtapiException.CTIERR_PERSISTENT_CALL_BEING_SETUP: Indicates that the request failed because a persistent call is already being set up.

The Play Zip Tone feature allows Cisco JTAPI application to play zip tones on active calls. The application specifies the type and the direction of the tone.

Zip tones are played at local or remote end of the call. They are audible and played only for IP phones. These tones are not played if the remote side is a trunk, conference or Cisco Media Terminal or Route Terminal.

The following tones can be played:

• CiscoTone.ZIPZIP

• CiscoTone.ZIP

• CiscoTone.CALLWAITINGTONE

Sample Code

Void playTone(TerminalConnection termConn,  int tone,  int direction){
      If ( termConn  ! =  null ){
try {
((CiscoTerminalConnection)termConn).playTone(tone,  direction);
} catch (Exception e){
System.out.println("Exception for playtone request " + e);
}

The presentation indicator (PI) on a call provides the application with the ability to hide or reveal Calling/Called/CurrentCalling/CurrentCalled/LastRedirecting parties name and number to the end user. JTAPI provides functions on CiscoCall to get PI value for the party. Use this PI info to present the parties information to the end user. These functions return a value of true or false. A value of "True" indicates that presentation in "Allowed, " and a value of "False" indicates the presentation is "Restricted."

For a conference call, the interfaces on CiscoCall do not return a correct value. Applications must iterate through all the connections in the call to get the PI value that is associated with the address for which the connection gets created. The interface that is provided on CiscoConnection is getAddressPI().

The following new interfaces exist on CiscoCall retrieve PI values.

The following interface on CiscoConnection retrieves the PI value for the address that is associated with the connection:

No change exist in the message flow.

This feature enhances the privacy of private held calls. When privacy is enabled, only the phone that placed a call on hold can retrieve that call, and the calling name and number are not displayed.

The feature provides the ability for a shared address to determine whether other shared addresses may barge into a call. When privacy is enabled, other shared address cannot barge into the call. Privacy is a terminals property. On IP phones, a Privacy feature button allows the users to enable and disable the privacy feature. Privacy can be dynamically enabled and disabled for the active calls on the terminal. When Privacy is on for a call, the TerminalConnection state available to other shared addresses is set to In Use. If Privacy status is changed during the CallProgress, CiscoTermConnPrivacyChangedEvent is delivered to the application.

In prior releases, if Privacy is enabled and the call is put on hold, all TerminalConnections were in TermConnHeld state and any other shared Address terminalConnection could unhold the call. In Cisco Unified Communications Manager 4.2, if the Enforce Privacy on Held Calls service parameter is enabled, and if Privacy is enabled for a call, putting the call on hold does not change the terminalConnections of other shared addresses and they remain in the In Use state.

If an outbound call is initiated through the API to an unallocated directory number across the European PSTN, the application will perceive the ConnFailedEv event with the cause as CiscoCallEv.CAUSE_UNALLOCATEDNUMBER. For the US PSTN, the application may not see any event.

To make the behavior consistent across the European and American PSTNs and also to address backward compatibility issues, a new service parameter UseProgressAsDisconnectedDuringErrorEnabled was added to the jtapi.ini file starting with JTAPI Version 1.4(3.21), which, when enabled (1 = enable; 0 = disable; the default is disable), causes applications to perceive ConnFailedEv in both cases.

QSIG Path Replacement, a network feature, optimizes the real-time protocol (RTP) path when calls are transferred or forwarded to other PBXs that are connected through QSIG trunks. When path replacement is in progress, a small window of time exists when the feature requests from applications would be ignored and JTAPI would throw an exception to the application.

The Global Call ID or the call is changed when the RTP path is optimized with a direct path between the starting terminating PBXs. JTAPI provides new interfaces to monitor the call.

QuietClear occurs at the other end when two parties are on a call, and one address goes out of service because of a network outage, the Cisco Unified Communications Manager goes down, the application controlling CTIPort goes down, or CTIManager goes down. At this stage, the other end of the call can only drop the call or disconnect the connection. It cannot perform any other callControl operations.

For the party that went out of service, applications will perceive ConnDisconnectedEv and/or TermConnDroppedEv, and the other end of the call receives ConnFailedEv with CiscoCause of CiscoCallEv.CAUSE_TEMPORARYFAILURE.

If applications try to invoke the following features during QuietClear mode, PlatformException with error code of CiscoJtapiException.CTIERR_OPERATION_FAILER_QUIETCLEAR gets thrown:

• Consult transfer

• Consult conference

• Blind transfer

• Hold

• Unhold

Note	Applications may only drop the call in this mode.

Whenever a media stream must be created between two endpoints, Cisco Unified Communications Manager issues start transmission and start reception events to both endpoints. In JTAPI, the CiscoRTPOutputStartedEv and CiscoRTPInputStartedEv events represent the start transmission and start reception events. The CiscoRTPOutputStartedEv.getRTPOutputProperties() method returns a CiscoRTPOutputProperties object, from which the application can determine the destination address of its peer endpoint in the call, as well as the other RTP properties for the stream such as payload type and packet size. Similarly, the CiscoRTPInputStartedEv.getRTPInputProperties() method returns a CiscoRTPInputProperties object that informs the application of the RTP characteristics for the inbound stream.

At any time while media is flowing, the current CiscoRTPOutputProperties and CiscoRTPInputProperties also remain available from the CiscoMediaTerminal.getRTPOutputProperties() and CiscoMediaTerminal.getRTPInputProperties() methods as well. These methods throw an exception if the CiscoMediaTerminal is not currently supposed to transmit or receive media.

When Cisco Unified Communications Manager wants the application to stop sending or receiving media as the result of a call disconnecting or being put on hold, for example, it sends the CiscoRTPOutputStoppedEv and CiscoRTPInputStoppedEv events. These events mean that the current RTP media stream that exists between the two endpoints should be torn down.

JTAPI 1.2 specifies that one of the preconditions of the CallControlConnection.redirect() method specifies for the state of the connection to be in either the CallControlConnection.OFFERING or the CallControlConnection.ALERTING state. Cisco Unified JTAPI also allows a connection in the CallControlConnection.ESTABLISHED state to get redirected.

The redirect() method includes the following overloaded form in the CiscoConnection interface. It allows applications to specify the behavior that is desired when a failure occurs while a call is redirected and specifying the calling search space, or resetting the original called field.

Applications choose the desired behavior, by passing one of the following INT parameters in the overloaded redirect method from the CiscoConnection interface:

• Redirect drop on failure—When a call is directed to a busy or an invalid destination, Cisco Unified Communications Manager can either drop the call if the redirect fails or leave the call at the redirect controller. The JTAPI application can then take corrective action, such as redirecting the call to another destination. The option for the redirect mode parameter follows:

  • CiscoConnection.REDIRECT_DROP_ON_FAILURE

  • CiscoConnection.REDIRECT_NORMAL

• Calling Address search space—Redirect uses the calling search space parameter to indicate which callingSearchSpace is used. Applications can either use the calling party search space or the redirect controller search space. The parameter options for this scenario follow:

  • CiscoConnection.CALLINGADDRESS_SEARCH_SPACE

  • CiscoConnection.ADDRESS_SEARCH_SPACE

• Resetting original called—The called address option parameter gets used to reset the original called fields. The options for this scenario follow:

  • CiscoConnection.CALLED_ADDRESS_UNCHANGED

  • CiscoConnection.CALLED_ADDRESS_SET_TO_REDIRECT_DESTINATION. This option affects the fields when the call arrives at the redirect destination.

For more information, refer to the com.cisco.jtapi.extensions.CiscoConnection documentation.

For the scenario where A Calls B, B redirects to C, and C (redirect destination) does not represent a provider observed address, JTAPI would provide CallCtlConnAlertingEv for C with cause code Ev.CAUSE_NORMAL. Prior to release 5.0, the cause code specified Ev.CAUSE_REDIRECT for the same scenario.

This change kept the behavior consistent for scenarios where C observed or did not observe the provider.

Note	When C is observed, for the same scenario, CallCtlConnAlertingEv at C is provided with CAUSE_NORMAL from releases prior to 5.0, and that behavior continues without change.

Cisco Unified JTAPI applications can specify the preferred original called party DN in the redirect request. The Redirect Set Original Called ID feature lets applications redirect a call on a connection to another destination while letting the applications set the OriginalCalledID to any value. This enables applications to transfer the call directly to the voice mail of another. For example, if A calls B and B wants to transfer the call to CVoice Mail, applications can specify in the enhanced redirect request C as the preferred original called party and destination party as CVoice Mail profile. With this request, calls appear in C Voice Mail profile with the Cisco Unified Communications Manager originalCalledParty field as C. Typical voice mail applications look for originalCalledParty information to identify a user voice mailbox.

Any application that redirects a call to a party by modifying the original called party can take advantage of this feature.

Note	This feature also changes the lastRedirectedAddress to the preferredOriginalCalledParty that gets specified in the redirect request.

The following callControlConnection interface applies for Redirect Set Original Called ID:

Interface CiscoConnection Extends callControlConnection With Additional Cisco Unified Communications Manager-Specific Capabilities

For details on the interface, see Cisco Unified JTAPI Extensions To view the message flow for Redirect Set Original Called ID, see Message Sequence Charts

Configuration requires that devices are configured into device pools and are assigned static Cisco Unified Communications Manager groups. Devices register with a particular Cisco Unified Communications Manager server that handles call control signaling. When a server fails, the devices failover to the backup server in the group. When the primary server comes back online, it waits until no active calls exist on the device, then re-homes to the primary Cisco Unified Communications Manager server. Cisco Unified JTAPI informs the applications of this transition by sending a temporary out-of-service message while registering to the backup server.

Cisco Unified JTAPI also offers transparent applications for redundancy via the CTI Manager. When the primary CTIManager fails, Cisco Unified JTAPI automatically connects to the backup CTI Manager and communicates the reconnection to applications. Instead of connecting to a single Cisco Unified Communications Manager server, applications now connect to a set of CTIManagers. The applications supply the CTIManager server names when they invoke JTAPI.

Cisco Unified JTAPI and the CTIManager maintain bidirectional heartbeat signals to detect a loss of connectivity between them. The CTIManager detects when an application no longer runs and cleans up its allocated resources. The following figure illustrates the“Logical Representation of JTAPI, CTIManager and Cisco Unified Communications Manager in a cluster”

Note	After Cisco Unified JTAPI successfully connects to the primary CTIManager, it alternately will attempt to reconnect to the primary or backup CTIManager if the JTAPI connection to the CTIManager fails.

In Release 11.0(1), Cisco JTAPI has been updated with how it responds to SIP 183 messages when a call is transferred over a gateway or trunk. When an established call gets transferred over a trunk and a SIP 183 is received, Cisco JTAPI moves the call to CallControlConnection.NETWORK_ALERTING state. When the call is answered, Cisco JTAPI moves the call state to CallControlConnection.ESTABLISHED.

A new Cisco CallManager service parameter, CTI Report Ringback on SIP 183 with SDP, has been added to configure this feature. When this sevice parameter is set to True, the above behavior applies. This is the default setting.

If an application needs to use the legacy behavior, you can set the service parameter to False. Under this setting, if the call call is transferred over a gateway or trunk, CTI will use the CallControlConnection.NETWORK_REACHED state to report that the other network has been reached, but CTI will not report back that a connection has been established.

Routing in JTAPI requires the configuration of a CTI Route Point on the Cisco Unified Communications Manager. Multiple calls can be queued to this Route Point, but only a single line can be configured on a CTI Route Point device.

JTAPI implementation of adjunct Routing, as described in the call center package, includes the following actions:

• Registering route callbacks on Route Addresses

• Creating appropriate handlers in response to the various routing events (routeSelect, routeEnd)

  Note

  CTI Route Points represent devices that can process any number of incoming calls simultaneously on the same line. You can route calls by using the methods in the javax.telephony.callcenter package, or you can accept, redirect, or disconnect calls by using the methods in the javax.telephony.callcontrol package. You can configure each CTI Route Point with a maximum of 34 lines. To support more than 34 lines, provision additional route points. For details on how to configure and administer the CTI Route Point, refer to the Cisco Unified Communications Manager Administration Guide.

The following figure shows the CTI Route Point configuration.

This feature informs applications whether a call is secure, allowing for secure conference calls. When the overall security status of the call changes, secure conferencing provides applications with a notification in the form of an event on the call. Applications receive the overall call security status of the call in the CiscoCallSecurityStatusChangedEv when the overall call security status changes. When a terminal goes to the talking state, JTAPI provides the call security status information to the applications. Applications can query the security status of the call by using a new interface on CiscoCall. The system makes the security status information available to applications when the applications start monitoring an existing call.

In shared address scenarios, the system also reports CiscoCallSecurityStatusChangedEv to the RIU parties. The OverallCallSecurityStatus matches the status reported on the active terminals. For example, in a three-party conference with A (Encrypted), B (Encrypted), C (Authenticated), and C' (Authenticated), the system reports CiscoCallSecurityStatusChangedEv with OverallCallSecurityStatus = Authenticated to C and C'. The system delivers this event on a per-call basis.

SRTP key information will continue to be sent for encrypted parties whether or not the OverallCallSecurityStatus is Encrypted. For example, in a three-party conference with A (Encrypted), B (Encrypted), and C (non-secure), the OverallCallSecurityStatus of the conference call is NotAuthenticated. However, the media that connects A, B, and the conference bridge continues to be encrypted because they are encrypted parties. Thus, A and B receive SRTP keys despite the OverallCallSecurityStatus.

This feature provides the mechanism that is needed to deliver Secure Real-Time Protocol (SRTP) key material of an encrypted media session between authenticated end points within Cisco Unified Communications Manager based Enterprise systems. To receive this key material, the administrator must configure the TLS Enabled and SRTP Enabled flags in the Cisco Unified Communications Manager Administrator windows and a TLS link must be established between JTAPI and the CTIManager.

Key materials get exposed in CiscoRTPInputKeyEv and CiscoRTPOutputKeyEv. To get these events, applications must enable rtpKeyEvenabled in CiscoTermEvFilter. By default, filters are disabled to maintain backward compatibility. If filters are enabled, application always get CiscoRTPInputKeyEv and CiscoRTPOutputKeyEv. A security indicator in these events indicates whether the media is encrypted and whether keys are available.

CiscoRTPInputKeyEv contains key material of the input stream and CiscoRTPOutputKeyEv contains key material of the output stream. Applications can use this key material to decrypt the packets and start monitoring or recording the media. Applications must not store this key material in a way that leaves the material vulnerable to tampering, and applications must zero out or clear the entry for these keys when they go out of scope.

This key material contains

• Key Length

• Master Key

• Salt Length

• Master Salt

• AlgorithmID

• isMKIPresent

• Key Derivation Rate

This enhancement also supports a secure media termination for CTIPorts and RoutePoints. To do this, the application passes in supported encrypted algorithms in CTIPort and route point register requests. The application gets an error if no TLS link and no SRTP Enabled flags exist. Whether media are encrypted or not depends on whether the other end is interested in secure media and whether the algorithm is negotiated successfully.

For mid-call monitoring, if the application comes up after a call is established between two end points, the application must query Terminal.createSnapshot() and snapshot event CiscoTermSnapshotEv. CiscoTermSnapshotCompletedEv gets sent, which indicates whether the current media between end points is secure or not. Applications can query CiscoMediaCallSecurityIndicator to get a security indicator for a call; however, this does not contain any key material in the event. If no calls exist on any of the lines on the terminal, applications only get CiscoTermSnapshotCompletedEv. To maintain backward compatibility, these events get generated only when an application enables the snapShotRTPEnabled filter in CiscoTermEvFilter.

CiscoRTPHandle gets added in all RTP events so that applications can correlate RTP events related to a single call. For backward compatibility, no new events are generated when there is no secure media.

For more information on SRTP, see the Secure RTP Library API Documentation by David McGrew on SourceForge.net.

The following sections describe the interface changes for SRTP key material.

This feature enables Cisco JTAPI to monitor and record secured calls. Monitoring and recording of calls was introduced in Release 6.0 of the Cisco Unified Communications Manager, but it did not support secured monitoring or recording of calls. For this release, the feature also supports secured calls. With this enhancement a supervisor or recorder can monitor or record a secure call only if its device security capability is same as or more than that of the agent. If the security capability of the monitor initiator's device is less than that of the target, the request for monitor fails. Recording request fails if the recording is attempted for an authenticated device, or if the security capability of the recorder is non-secured and that of the agent is Encrypted.

Cisco JTAPI throws a PriviledgeViolationException with CTIERR_SECURITY_CAPABILITY_MISMATCH, when the monitoring request is rejected due to the supervisor not meeting the security capabilities of the agent. A new API getTransactionID() is added to CiscoTermConnMonitorInitiatorInfoEv and CiscoTermConnMonitorTargetInfoEv.

CiscoJTAPI delivers a new event CiscoAddrMonitoringTerminatedEv when the monitoring session is torn down. This event is delivered to the Supervisor who had started the secured monitoring session but had dropped off from the monitoring call.

New APIs getCiscoAddrMonitoringTerminatedEvFilter() and setCiscoAddrMonitoringTerminatedEvFilter() have been added to the interface CiscoAddrEvFilter for applications to get or set the filter value for the CiscoAddrMonitoringTerminatedEv. By default, the filter is set to True and the event is delivered. To stop receiving this event, applications must set this filter to False.

As before, When a monitoring call (call used by monitor initiator) is conferenced, the final call may not have any connection to monitor target. When monitor initiator conferences another party to a monitoring call, both parties can to listen to the audio between monitor target and caller.

The SelectRoute interface gets enhanced to take the parameters PreferredOriginalCalledNumber and PreferredOriginalCalledOption. This enables applications to reset the OriginalCalled value to a specified PreferredOriginalCalledNumber when the call gets routed. This interface takes a list of PreferredOriginalCalledNumber, PreferredOriginalCalledOption, and corresponds them to the RouteSelected list. If the call gets routed to Route at index I in the RouteSelected list, the PreferredOriginalCalledNumber and PreferredOriginalCalledOption at index I get used. Applications get the following behavior with different values for these parameters.

Note

Below x, point to the index where the call is being routed. For example, if the call gets routed to Route n, then value of x will equal n. If a PreferredOriginalCalledOption at index x is invalid or out of range, JTAPI defaults it to CiscoRouteSession.DONOT_RESET_ORIGINALCALLED, and if PreferredOriginalCalledOption is null, all the routing gets done with option CiscoRouteSession.DONOT_RESET_ORIGINALCALLED.

Note	When OriginalCalled gets set to PreferredOriginalCalled, LastRedirectingParty number also gets reset to PreferredOriginalCalled.

The following new or changed interfaces exist for SelectRoute Interface Enhancement:

PreferredOriginalCalledOption takes one of the following values:

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for SelectRoute Interface Enhancement, see Message Sequence Charts.

The selectRoute() has feature priority and calling search space parameters as an array. This API provides the flexibility of different feature priorities and calling search spaces for each route selected.

SetMessageWaiting provides a method for applications to set the message-waiting lamp or indicator for an address. Invoke the method on an address that is in the same partition as the destination.

The following interface specifies whether the message waiting indicator should be activated or deactivated for the address that the destination specifies. If enable is true, message waiting activates if not already activated. If enable is false, message waiting deactivates if not already deactivated.

{
public void setMessageWaiting (java.lang.String destination,  boolean enable)
   throws javax.telephony.MethodNotSupportedException, 
          javax.telephony.InvalidStateException, 
          javax.telephony.PrivilegeViolationException

}

Shared line represents the same DN appearances on multiple terminals. CiscoJtapi provides support for Shared Line, which provides applications with the ability to control shared DN terminals, hold a call on one shared DN Terminal and unhold the same call from another shared DN Terminal, make calls between two shared lines, initiate a call from one shared line terminal while another active call exists on another shared line terminal with the same DN.

Share line provides the following interfaces:

• CiscoAddress.getInServiceAddrTerminals()—Returns an array of terminals for which the address is in service.

  Terminal {} getInServiceAddrTerminals();

• CiscoAddrOutOfService.getTerminal()—Returns the terminal that is going out of service.

  Terminal getTerminal();

• CiscoAddrInService.getTerminal()—Returns the terminal that is going in service.

  Terminal  getTerminal();

• CiscoConnection.setRequestController(TerminalConnection tc)—Allows an application to select a terminalConnection that is associated with a connection on which you can perform park, redirect, or disconnect operations. You need to do this in a situation where more than one active TerminalConnection exists in a SharedLine scenario.

• CiscoConnection.getRequestController()—Returns TerminalConnection that application sets as request controller.

• CiscoAddrAddedToTerminalEv—Gets sent when the following conditions occur:

  • A Terminal/Device gets added into the user controlList that contains a SharedDN, which sends the event to the application. In other words, if user has an address in control list, and a new device gets added with same address in control list, this event gets sent.

  • An EM (extension mobility) user logs into the terminal with a profile that contains a SharedDN. In this scenario, this event notifies that a new terminal is added to an already existing Address.

  • A new SharedDN is added to a device in a user control list

  Interface getTerminal() returns the terminal that gets added to the address.

  Interface getAddress() returns the address on which a new terminal is added.

• CiscoAddrRemoveFromTerminalEv—Gets sent when the following conditions occur:

  • A user removes a Terminal/Device from the user controlList that contains a SharedDN. In other words, if a user has a shared address in a control list, and one of the devices with same address gets removed, this event gets sent.

  • An EM(extension mobility) user logs out from the terminal that had a profile that contains a SharedDN. This event notifies applications that one of the terminals is removed from an existing Address.

  • A new SharedDN (SharedLine) is removed from a device in a control list.

  Interface getTerminal() returns the terminal that gets removed from the address.

  Interface getAddress() returns the address from where the terminal gets removed.

The following changed or new behaviors exist for a SharedLine:

• Behavior changes for CiscoAddress event include

  • JTAPI applications will receive multiple CiscoAddrInServiceEv for shared line addresses. Applications can use CiscoAddrInServiceEv.getTerminal() to get the terminal on which address goes in service.

  • JTAPI applications receive multiple CiscoAddrOutOfServiceEv for shared line addresses. Applications can use CiscoAddrInServiceEv.getTerminal() to get terminal on which address goes out of service.

  • The address state goes in service when a first shared line goes in service; for example, when the first CiscoAddressInServiceEv gets received.

  • The address state goes out of service when the last shared line goes out of service; for example, when the last CiscoAddressOutOfServiceEv gets received.

• For an incoming call, all the line appearances of a shared line ring. To applications, this gets presented as one active call (callActiveEv), one Connection(ConnCreatedEv), and multiple terminalConnection(TermConnCreatedEv one each for each shared line).

• Calls get presented to all terminals. When a call is in a ringing state, the state of the terminal connection equals Ringing. When a the shared line answers, the terminalConnection state goes to an active state, while other terminalConnections on the shared line go to a passive state, and callControlTerminalConnection for all the shared lines at this point go into a bridged state. When a call is put on hold, all the terminal connections go into an active state, and callControllTerminalConnection goes to a held state. At this point, any terminal can retrieve the call. The retrieving terminal terminalConnection remains in an active state, and callControlTerminalConnection goes to a talking state while all other shared terminals terminalConnections go into a passive state. Simultaneously, CallControlTerminalConnection changes from a held state to a bridged state.

• A shared line can make a call to another shared line of the same DN. In this scenario, the call includes only one connection and multiple terminal connections.

• When a shared line makes a call to another shared line of the same DN, the post condition for this equals only one connection.

• For a shared line connection with two active terminalConnections (such as barge), Connection.Disconnect() does not result in disconnected connection.

  If an application is monitoring only a SharedDN Connection with only a passive or bridged TerminalConnection, invoking any API on the connection results in a PreConditionException.

• Similar to the previous scenario, if all the connections of a call monitored by an application have only a Passive or Bridged TerminalConnection, all APIs on the call throw a PreConditionException (such as Call.Drop()).

• If more than one active TerminalConnection exists on a shared line, Call.drop() does not return in CallInValid in the following scenarios:

  • A normal two-party call between A and B, where A represents a SharedLine with A' and A' barged into the call

    The application does not monitor A' and B. If the application issues a Call.drop(), the A’ TerminalConnection goes into a passive state, but the call does not go InValid.

  • Similar to above, if A, A' , A" and B are in a Conference Call

    The application monitors only A and A', and Call.drop() does not result in the call going InValid. Only the A and A' terminal connections go passive.

  • A, A', and B, B' represent a SharedLine address

    A calls B, B answers, and A' and B' barge into the call. The application monitors only A and B. In this scenario, Call.drop() results in a TerminalConnection of A and B going passive, but the call does not go InValid.

• If a TerminalConnection is in a passive or bridged state or Passive/InUse state, all APIs on the TerminalConnection() throw a PreConditionException. A TerminalConnection only allows an API Terminal ConnectionJoin() (called Barge) in the passive or bridged state. TerminalConnection does not currently support TerminalConnection Join().

• If more than one active or talking TerminalConnections exists in a connection, applications may have to end one before issuing an API on the connection like Redirect(), Park(), Disconnect(). You can select TerminalConnection by using API Connection.setRequestController (TerminalConnection tc).

• If a call gets held on SharedLine terminals and an application issues a Connection.Disconnect (), the applications may set a particular TerminalConnection through API Connection.setRequestController(TerminalConnection tc). If requestcontroller is not set, all HeldTerminalConnections get dropped, and connection goes to a disconnected state. If only one HeldConnection gets dropped, the call remains present on other SharedLines terminals. The call appearance disappears from the dropping terminal, which disallows the terminal from barging into the call or participating in feature operations on the call.

For details on the interface changes, see Cisco Unified JTAPI Extensions To view the message flow for shared lines, see Message Sequence Charts

This feature provides the ability to silently monitor calls using an IP Phone. The caller represents the end point, which calls or receives a call from the monitor target. The monitor target is the party to monitor (in a call centre, the agent), and the monitoring party is the monitor initiator (the supervisor).

The recording feature lets applications record conversations on any observed address. Three recording configurations are available:

The silent monitoring feature lets applications listen to a live conversation between two other parties. The monitor initiator cannot talk to either the monitor target or the caller. The feature provides notification tones when legal compliance is required.

Only an application request can initiate monitoring. The application must send a monitor request for each call that it wants to monitor. The system can only monitor calls that are in a connected state. On the successful completion of a monitor request, the audio stream between the monitor target and the caller streams to the monitor initiator. The monitor target receive a tone:

• if the monitor target is configured to receive a tone, or

• if the application requests a tone when it starts the monitor

Applications can monitor calls if they belong to the Standard CTI Allow Call Monitor user group or can be used outside of contact center. The system delivers monitoring-related events to all call observers.

"Monitor" is a reserved word that should not be configured as display names for any lines in the system. Other reserved words are "Conference, " "Park Number, " "Barge, " and "CBarge."

When a monitoring session is established, the terminal observer on the monitoring initiator receives Cisco RTP events. Although the media for a silent monitoring call flows only in one direction, getMediaConnectionMode() would return CiscoMediaConnectionMode.TRANSMIT_AND_RECEIVE instead of CiscoMediaConnectionMode.RECEIVE_ONLY. Applications should expect to find the same behavior in CiscoMediaOpenLogicalChannelEv if a CTIPort is used as the monitor initiator.

When a monitoring call (the call used by the monitor initiator) is conferenced, the final call does not have any connection to the monitor target. When the monitor initiator conferences another party into a monitoring call, both parties can listen to the audio between the monitor target and the caller.

The following interfaces extend TermConnEv and are delivered to the call observer. For shared lines, the system delivers these events to call observers on the address or terminal of the talking terminal connections. Applications receive no events if they have only the terminal whose connection is in the INUSE or BRIDGED state.

CiscoTermConnMonitoringStartEv

CiscoTermConnMonitoringStartEv

Indicates the start of monitoring and is delivered to the call observer on the monitor target. Using getMonitorType() on this event returns the monitor type.

CiscoTermConnMonitoringEndEv

CiscoTermConnMonitoringEndEv

Indicates the end of monitoring and is delivered to the call observer on the monitor target.

CiscoTermConnMonitorInitiatorInfoEv

Exposes monitor initiator information and is delivered to the call observer of the monitor target. This interface has one method:CiscoMonitorInitiatorInfo getCiscoMonitorInitiatorInfo ()

Returns a CiscoMonitorInitiatorInfo that exposes the terminal name and address of the monitor initiator.

CiscoTermConnMonitorTargetInfoEv

Exposes monitor target information and is delivered to the call observer of monitor target. This interface has one method:CiscoMonitorInitiatorInfo getCiscoMonitorTargetInfo()

Returns a CiscoMonitorInitiatorInfo that exposes the terminal name and address of the monitor target.

Two new error codes notify applications about monitoring failures:

• CTIERR_PRIMARY_CALL_INVALID is returned by CiscoException.getErrorCode() for exceptions that occur when a monitoring request fails due to the call going idle or getting transferred.

• CTIERR_PRIMARY_CALL_STATE_INVALID is returned when the monitoring request fails due to the call transitioning to a different state where monitoring cannot be invoked.

This release introduces a new AddressType, MONITORING_TARGET. JTAPI creates a connection on an address of this type for a monitoring target address; CiscoAddress.getType() returns this value.

The Single Sign-On feature allows Cisco JTAPI applications to use the single sign-on ticket to authenticate instead of a user ID and password.

Applications fetch the service ticket for the OpenSSO server from the active directory and then pass the ticket to Cisco JTAPI in the string used in the getProivder(String str) API. Applications can set the single sign-on ticket as ssoticket = "ssotokenfromat".

Only end users can use this feature.

Applications using this feature need not specify the user ID and password in the getProvider string.

If an application is used by an end user and has the Standard CTI Secure Connection role enabled, then a user ID is required in the provider string. No password is required.

This solution is designed around an active directory with a Kerberos environment to achieve Windows desktop Single Sign-On. If an active directory with a Kerberos environment is unavailable, then an alternate equivalent setup is available, which includes a KDC, an authentication server, and a domain controller.

Sample Code

String ssoticket  =  getSSOticket(); //application implementation
String providerString  =  cucmserver + ssoticket +";";
JtapiPeer peer  =  JtapiPeerFactory.getJtapiPeer ( null );
try
{
  Provider provider  =  peer.getProvider (providerString);
}
catch (Exception exp )
{
  if ( exp instanceof PlatformException)
  {
    switch (((CiscoJtapiException)exp).getErrorCode() )
    {
    case CiscoJtapiException. CTIERR_SSO_DISABLED:
      System.out.println("SSO feature not enabled on CUCM ");
      break;
    case CiscoJtapiException. CTIERR_SSO_AUTH_SERVER_DOWN:
      System.out.println("server down");
      break;
    }
  }
  else
  {
    System.out.println("Exception  =  " + exp.toString());
  }

}      

This interface allows applications to transfer a call to an address. Cisco Unified JTAPI continues to support this interface as defined in JTAPI 1.2 specification, but the events that are delivered to applications are changed from the previous versions of Cisco Unified JTAPI.

In previous versions of Cisco Unified JTAPI, the original call goes to a held state, and a new call gets created between the transfer controller and destination when applications use this interface. After successful completion of transfer, both calls on transfer controller go to an IDLE state. If a transfer fails, the original call remains in a held state, and applications retrieve the call. CiscoTransferStart and end events get delivered to the applications at the start and completion of the transfer operation.

Applications get the following changes:

• A new call does not get created.

• CiscoTransferStartEv and CiscoTransferEndEv do not get delivered to applications.

• The state of the original call is retained if the transfer operation fails.

The pre and post conditions of this interface did not change.

To view the message flow for Single Step Transfer, see Message Sequence Charts

The SIP Redirect server receives SIP requests and responds with 3xx(redirection) responses, which direct the client to contact an alternate set of SIP addresses. This enhancement supports the Cisco Unified Communications Manager Redirection (3xx) Call Control primitive in compliance with RFC 3261. The Cisco Unified Communications Manager Redirection primitive processes SIP 3xx responses and does sequential hunting to each contact address from the 3xx response. Cisco Unified Communications Manager Redirection primitive also handles feature interactions that result from performing this operation. Cisco Unified JTAPI exposes new reason codes in all CallEvs, which indicate when connection and terminalConnection are created and destroyed as a result of this primitive.

LastRedirectAddress may change if feature interactions like JTAPI Redirect or CallForwardNoAnswer occur when the Redirection primitive is hunting for a target. If the target does not answer and Cisco Unified Communications Manager Redirect takes control of the call to send it to next target, lastRedirectAddress is set to the party who originally sent the SIP 3xx response.

If a diversion header is present in the SIP 3xx response, the 3xx primitive uses the first value of the diversion header for lastRedirectParty, and JTAPI applications will see the diversion header element as lastRedirectAddress.

To maintain backward compatibility, JTAPI exposes the new API CiscoCallEv.getCiscoFeatureReason() in the CiscoCallEv interface, which contains the reason as CM_REDIRECTION.

Note	Applications should be aware that new feature-specific reason codes could be returned from this API, and applications should provide default behavior for unrecognized reason codes.

The following sections describe the interface changes for SIP 3XX Redirection.

This release of Cisco Unified Communications Manager allows phones that run SIP to register and interoperate with phones that run SCCP. The following sections describe the new interfaces introduced to support phones that run SIP along with the limitations and differences in behavior with respect to phones that support SCCP. Though not all existing features are supported on phones that run SIP, the general behavior in terms of JTAPI events and interfaces for phones that run SIP are similar to that of a phone that runs SCCP.

JTAPI applications can only control Cisco Unified IP Phone 7900 Series that run SIP, which includes Cisco Unified IP 7970 phones. Applications should not include Cisco Unified IP 7960, 7940, and other phones that run the SIP protocol in their control list. JTAPI applications cannot control third-party phones that run SIP, so third-party phones that run SIP should not be included in the control list.

In prior releases, JTAPI supported an initial feature set on phones that run SIP. In this release support is added for the following functionality on phones that run SIP:

• Park for Phones that run SIP

• Unpark Phones that run SIP

Tip	The order of events for consult calls differs for phones that run SIP and SCCP phones. Consider the following scenario:

1. Terminal A initiates a call to the shared line B/B'.
2. The shared line initiates a consult call to Terminal C.

   If the shared line is a SIP device, the following call events occur:

   • B (active) receives: OnHold -> Select -> NewCall

   • B' (remote-in-use) receives: Select -> NewCall -> OnHold

   However, if the shared line is a SCCP device, the call events are Select -> OnHold -> NewCall on both terminals.

   If the application is only monitoring, call.getConsultingTerminalConnection() may return null.

JTAPI supports the following features for phones that run SIP:

• Call.connect; offhook

• answer; disconnect; drop; hold, unhold

• consult; transfer; conference; redirect

• playdtmf, deviceData

JTAPI supports the following events for phones that run SIP:

• CiscoTermDeviceStateEv, RTP events, inService, and OutOfService

• MediaTermConnDtmfEv (only out of band is supported), transfer start and end events, conference start and end events, CiscoToneChangedEv, and CiscoTermConnPrivacyChangedEv

Behavior of phones that run SIP differ from that of phones that run SCCP in the following ways:

• Call Rejection—When a call is made to a phone that runs SIP, the phone can choose to reject the call. In this case, applications perceive CallActive, ConnCreatedEv followed by ConnDisconnectedEv for the address on the SIP terminal. This is similar to RP rejecting the call.

• Consult without media calls involving SIP phones should be transferred within 1.5 seconds after the call is connected.

• For phones that run SIP, enbloc dialing is always used even if the user first goes off hook before dialing digits. The phone waits until all the digits are collected before sending the digits to the Cisco Unified Communications Manager . This means that CallCtlConnDialingEv is delivered only after enough digits are pressed on the phone to match one of the configured dialing patterns.

• Applications should configure "out of band DTMF" on all devices to receive MediaTermConnDtmfEv.

Events for CTI ports, route points, and phones that run SCCP are not changed.

When a Cisco Unified IP Phone 7900 Series model that runs SIP using UDP as transport fails connectivity with Cisco Unified Communications Manager , JTAPI applications receive the events CiscoTermOutOfServiceEv and CiscoAddrOutOfServiceEv for the terminal and address defined for the phone. Because of the inherent delay in UDP in detecting the connectivity loss, the Cisco Unified IP Phone 7900 Series that runs SIP may visually show as registered after applications have already been notified with the out-of-service events.

If Cisco Unified IP Phone s 7960, 7940, and non-Cisco Unified IP Phone 7900 Series that run SIP are included in the control list, exceptions are thrown when observers (both observer and call observers) are added to the address or terminal and CiscoTermRestrictedEv is delivered to a provider observer. The cause for these events would be CiscoRestrictedEv.CAUSE_UNSUPPORTED_PROTOCOL.

CiscoTerminal exposes new interface getProtocol() to indicate whether terminal is a phone that runs SCCP or a phone that runs SIP. CiscoTerminalProtocol defines the values that are returned by getProtocol().

The following new interfaces that are defined on CiscoCall let applications get URL information for external SIP entities.

REFER is a SIP method that is defined by RFC 3515. The REFER method indicates that the recipient (referee, identified by the Request-URI) should contact a third party (referred to as the target) by using the contact information that is provided in the request. This REFER method allows the party who is sending the REFER (referrer) to be notified of the outcome of the referenced request.

Cisco Unified Communications Manager, being a Back-To-Back User Agent (B2BUA), processes both inside and outside dialog inbound REFER on behalf of the Referee. As result of REFER, Cisco Unified Communications Manager creates a call between the Referee and the Refer-to-Target. Ifthere is a previously existing call between the Referrer and the Referee, the call at the Referrer gets dropped after REFER completes.

The REPLACES feature is the replacement of an existing SIP dialog with a new dialog. A SIP dialog is a call between two SIP user agents; a Cisco Unified Communications Manager dialog is a half call (callleg). The REPLACES feature is triggered either by REFER or by an INVITE. Cisco Unified Communications Manager handles a REPLACES request on behalf of the recipient of the REPLACES header. The request is associated with a new dialog and the requesting party is the party that wants to replace another party in the existing dialog (call) identified in the REPLACES header. Cisco Unified Communications Manager disconnects the dialog (call) identified in the REPLACES header and connects the requesting party.

JTAPI is enhanced to model Call events caused by the Cisco Unified Communications Manager REFER and REPLACE features in the JTAPI call model. JTAPI provides applications with the capability to handle call events caused by REFER and REPLACE features. JTAPI does not provide any interface for applications to initiate REFER or REFER/INVITE with REPLACES requests; however, JTAPI can handle the call events properly.

These two features are backward compatible. JTAPI provides events that are caused by REFER/REPLACE with CAUSE_NORMAL. Applications can get feature-specific reasons from the new interface CiscoCallEv.getCiscoFeatureReason().

Note	This interface provides feature-specific reasons for current and new features, but this method will not remain backward compatible in future releases. Applications using this interface must implement default handling to avoid future backward-compatibility issues.

The following sections describe the interface changes for SIP REFER/REPLACE.

com.cisco.jtapi.extensions
Interface CiscoCallEv

The SIP Trunk Early Offer feature allows the SIP trunk to support early offer outbound calls. The SIP trunk does not use a Media Termination Point (MTP) when the media capabilities and port information of the phone is available.

If the media port information is not available, the Cisco Unified Communications Manager allocates an MTP to provide an offer.

If the application enables this feature and makes a call that goes though a SIP trunk, the Cisco Unified Communications Manager must have the IP address and the port information of the registered terminal even before the media is established. This eliminates the need for MTP.

The following are the changes done from the JTAPI perspective:

• A new interface, CiscoBaseMediaTerminal, extends CiscoTerminal.

• A new register() API has the following arguments:

  • IP Address

  • Port

  • Media Capability

  • Algorithm ID

  • IP_V6 Address

  • Addressing Mode

  • Registration Type

Applications use register() API to register CiscoMediaTerminal and CiscoRouteTerminal with the following registration types available in CiscoBaseMediaTerminal.

• CiscoBaseMediaTerminal.NO_MEDIA_REGISTRATION (applicable only for route points)

• CiscoBaseMediaTerminal.DYNAMIC_MEDIA_REGISTRATION (for dynamic registration of CTI ports and route points)

• CiscoBaseMediaTerminal.DYNAMIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT

• CiscoBaseMediaTerminal.STATIC_MEDIA_REGISTRATION (for static registration of CTI port)

• CiscoBaseMediaTerminal.STATIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT

Note	The applications use the register() APIs on CiscoRouteTerminal and CiscoMediaTerminal for route points and CTI ports to specify the registration type.

To enable this feature, select one of the following:

• CiscoBaseMediaTerminal.DYNAMIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT for registration type to register a CTI port or a route point dynamically

• CiscoBaseMediaTermial.STATIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT for registration type to register a CTI port or a route point statically.

If an application has enabled this feature and initiated a call that goes through a SIP Trunk, CiscoJTAPI delivers a new event CiscoMediaOpenIPPortEv. On recieving this event, applications query for the registration type using the API getRegistrationType(), which is exposed on this interface, and do the following based on the value returned.

• If return value is CiscoBaseMediaTerminal.DYNAMIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT, applications must set the RTP Parameters and open the port. At present, the applications set the RTP parameters upon receiving CiscoMediaOpenLogicalChannelEv for dynamically registered CiscoMediaTerminal and CiscoRouteTerminal.

• If return value is CiscoBaseMediaTerminal.STATIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT, applications must open the port. At present, most of the applcations open statically registered terminals when they receive RTP events.

If an application tries to register a terminal, which is already registered with registration type as CiscoBaseMediaTerminal.DYNAMIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT or CiscoBaseMediaTerminal.STATIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT, with a different registration type, JTAPI throws a PlatformException with the error code as CiscoJtapiException.CTIERR_MEDIA_ALREADY_TERMINATED_DYNAMIC_GETPORT_SUPPORT or CiscoJtapiException.CTIERR_MEDIA_ALREADY_TERMINATED_STATIC_GETPORT_SUPPORT, respectively.

A new API, isRTPRequired(), is also exposed on the interface CiscoMediaOpenLogicalChannelEv to indicate if the applications must set the RTP parameters or not when they receive this event.

Applications must check the API when they recieve the CiscoMediaOpenLogicalChannelEv and set the RTP Parameters only when the return value is true.

Note	Early offer is not supported for IPv6 calls in release 8.5(1).

If an application registers a terminal with registration type as CiscoBaseMediaTerminal.DYNAMIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT or CiscoBaseMediaTerminal.STATIC_MEDIA_REGISTRATION_FOR_GET_PORT_SUPPORT and the IP addressing mode as IPv6, the registration follows but this feature does not come into effect. The applications do not receive the CiscoMediaOpenIPPortEv.

The application must close the ports when it receives media termination events or when a call is disconnected.

When IPv6 support is added, the application receives two CiscoMediaOpenIPPortEv, for dual mode devices, one for IPv4 and the other for IPv6 addresses. When the call is answered, application closes the unused port based on MediaIPAddressingType in CiscoMediaOpenLogicalChannelEv.

The service parameter, Fail Call Over SIP Trunk if MTP Allocation Fails, decides if the call must go through as a delayed offer or not. If applications do not set the RTP parameters when they receive CiscoMediaOpenIPPortEv for a dynamically registered terminal with get port support, this service parameter decides if the call must go through as a delayed offer or not.

The Star (*) 50 feature enables you to divert a call to original called party (value returned by CiscoCall.getCalledAddress() method) and the called party (value returned by CiscoCall.getCurrentCalledAddress() method) from phone UI. After pressing the iDivert softkey, a menu displays that identifies the names of the original called party and the called party.

The user selects one of the two names and the call is redirected to the voice mailbox of the selected party. With the legacy iDivert, the call is diverted to original called party voice mailbox by just pressing iDivert softkey. Cisco Unified Communications Manager Administration introduced the following Service parameters to configure this feature:

• iDivert Legacy Behavior—Determines whether the phone uses the legacy iDivert behavior when a user presses the iDivert softkey or the enhanced *50 iDivert behavior. If the iDivert legacy service parameter is set to true, the iDivert legacy behavior is adopted and vice versa.

• Allow QSIG during iDivert–Determines whether iDivert legacy is allowed in deployments that have voice messaging integration over QSIG trunks and only used when the Use Legacy iDivert service parameter is set to true.

• iDivert User Response timer–Determines the number of seconds that Cisco Unified Communications Manager Administration waits for a response from the user before the iDivert screen is removed. If no user action occurs by the time this timer expires, the screen is removed from the phone. If the Use Legacy iDivert service parameter is set to true, Cisco Unified Communications Manager Administration ignores this parameter.

There is no interface change at JTAPI layer for this feature. The behavior changes from JTAPI application point of view means that Calls could either go to voice mail of OrigicalCalled Party or Called.

When a JTAPI application user is configured, the system administrator normally associates a certain set of terminals (Cisco Unified IP Phones and devices) with this application user, who can control and monitor only this set of terminals. The Super Provider feature gives applications the ability to control and monitor any terminal in a Cisco Unified Communications Manager cluster.

The new createTerminal() new interface in CiscoProvider lets the application create a terminal by specifying a terminalName. JTAPI does not provide the capability to get the terminalName through any interface. The CiscoProvider.createTerminal(terminalName) returns the terminal. If the terminal already exists in the provider domain, JTAPI returns the existing terminal.

A second new interface, CiscoProvider.deleteTerminal(), lets the application delete the CiscoTerminal objects that are created by using the CiscoProvider.createTerminal() interface. If the terminal object does not exist or the application did not create the terminal with the CiscoProvider.createTerminal() interface, JTAPI throws exceptions.

JTAPI also provides a new interface on CiscoProviderCapabilities, canObserveAnyTerminal(), which can be enabled for application users through Cisco Unified Communications Manager Administration user configuration. Applications can use this interface to determine whether they have sufficient capability to invoke the createTerminal(terminalName) interface. If the application does not have sufficient capability and this interface is invoked, JTAPI throws a PrivilegeViolationException. If the application provides a terminalName that does not exist in the Cisco Unified Communications Manager cluster, JTAPI throws a InvalidArgumentException.

Superprovider enhancements for JTAPI in this release consist primarily of the following changes.

When the "Superprovider privilege" gets disabled from Cisco Unified Communications Manager Administration after a provider opens, JTAPI gets notified through a CTI Change Notification Event and cleans up all the devices that it has opened that are not in its control list.

JTAPI informs applications about the change using the "CiscoProviderCapabilityChangedEvent." This new event gets issued when the flag changes and indicates whether the flag has been enabled or disabled. When a device that is not in the control list is opened in the Superprovider mode, then moved to the control list, JTAPI moves the device into its control list.

• When a normal application receives a "CiscoProviderCapabilityChangedEvent" with the flag set, it means the Superprovider privilege has been granted to it, and it can start acquiring devices not in its control list.

• When a Superprovider application receives a "CiscoProviderCapabilityChangedEvent" with the Superprovider flag not set, it means that the Superprovider privilege has been removed for it. The following sequence of events then occurs:

  • Applications receive a Provider OOS event and all devices acquired/opened by it are closed.

  • Applications receive a CiscoTermRemovedEv for all devices not in the control list that have been acquired or opened.

  • Applications receive a Provider inService event when JTAPI succeeds in reconnecting to CTI as a normal user.

  • Applications receive device and line information.

  • Applications receive CiscoTermCreatedEv for all controlled devices that were open before the provider went OOS.

• JTAPI notifies applications by using the "CiscoProviderCapabilityChangedEvent" when the "park DN monitoring" flag is changed from Cisco Unified Communications Manager Administration.

  • When an application receives this event with the flag set, it does a register feature for the controlling park DN.

  • When an application receives this event with the flag not set, JTAPI again informs applications by using a "CiscoProviderCapabilityChangedEvent" and closes all the park DN addresses.

• JTAPI notifies applications by using the CiscoProviderCapabilityChangedEvent” when the “change calling party number” flag is changed from Cisco Unified Communications Manager Administration.

  • When an application receives this event with the flag set, it can change the calling party number.

  • When an application receives this event with the flag not set, it cannot change the calling party number.

  Applications should not change the calling party number when this flag is disabled.

• When a device that is not in the control list is opened or acquired by Superprovider, and is then deleted from Cisco Unified Communications Manager Administration, JTAPI closes the terminal object and sends a CiscoTermRemovedEvent to the application for that device.

public interface CiscoProviderCapabilityChangedEv {
    public CiscoProviderCapabilities getCapability ();
}

  public boolean canModifyCallingParty();
  public void setCanModifyCallingParty(boolean value);

Cisco Unified IP Phone 6901 is a new IP phone with keypad similar to other basic Cisco IP phones but this phone does not have display, speaker phone, or head set jack. This phone supports only SCCP protocol. Features such as Park, Unpark, Call Pickup, Group Call Pickup, Direct Transfer, Call Forward All, and Join are not supported as softkeys are not provided for these features. These features are supported only from Cisco Unified Communication Manager. Cisco Unified IP Phone 6901 is a one line device and can support two calls per line. So, features such as Join Across Lines and Direct Transfer Across Lines cannot be supported by these devices.

One of the limitations of this phone is that to intiate or answer a call, the phone must be off-hook. If the phone is on-hook and the user initiates or answers a call, JTAPI throws InvalidStateException to the application with error code as CiscoJtapiException.OPERATION_NOT_AVAILABLE_IN_CURRENT_STATE.

Another limitation is that Cisco Unified IP Phone 6901 does not accept XSI objects from applications, but if the application calls sendData() API for these phones, JTAPI throws an exception for the request to the application with the error code as CiscoJtapiException.COMMAND_NOT_IMPLEMENTED_ON_DEVICE.

This feature allows Cisco Unified JTAPI applications to control terminals with rollover mode enabled. In rollover mode, terminals are configured with multiple addresses with the same DN but in different partitions or with different DNs. When rollover mode is enabled, consult calls can be created on the next available address on the terminal. Cisco Unified IP Phone 6900 Series can be configured with rollover mode.

A new role Standard CTI Allow Control of phones supporting rollover mode has also been introduced to allow applications to control terminals with rollover enabled. Applications that support this new behavior where consult calls are created on a different address, must include this role to their application or end user. If not, all terminals configured with rollover mode are restricted and exceptions are thrown to addObserver() requests.

Applications that support this behavior are must add call observer on the terminal or add call observers on all addresses on the terminal. Since consult call is created on the next available addresses, exceptions are thrown to consult requests if call observers are not added to all addresses.

Join across lines must be enabled on Cisco Unified IP Phone 6900 Series to successfully complete conferences from applications.

Cisco Unified Communications Manager Release 8.6, JTAPI supports multiple calls per line configuration on Cisco Unified IP Phone 69xx series. Prior to Release 8.6, Cisco Unified IP Phone 69xx series supported only one call per line, where Maximum Number of Calls/Busy Trigger defined for a line (MNC/BT) cannot exceed 2/1. With multiple calls per line, Cisco Unified IP Phone 69xx series supports more than one call per line, and MNC/BT is configured to values greater than 2/1.

This feature enables users to have more than 100 Directory Numbers associated with a Device (Phones, CTI Ports and Route Points). JTAPI supports this feature and displays the corresponding addresses on the terminal to the application.

This feature enables Cisco Unified JTAPI support for Swap and Cancel operations on supported IP phones.

When a Swap operation is invoked, it puts an active call on hold and retrieves the held call. When a Cancel operation is invoked, it breaks the consulting relationship between primary and consulting calls. These operations can only be invoked from supported phones. The Cisco Unified JTAPI interface does not allow SWAP/CANCEL operations to be invoked from the application. Whenever a user presses the SWAP key on a phone, JTAPI delivers CallCtlTermConnHeldEv and CallCtlTermConnTalkingEv for active and held calls and indicates their state change with CiscoFeatureReason.REASON_NORMAL.

When a CANCEL operation is invoked and the relationship is broken between primary and consulting calls, Cisco Unified JTAPI is still able to use the Direct Transfer or Join feature to complete the transfer or conference operation. If the user presses the CANCEL key on phone after initiating a consult, the conference or transfer is not completed. Pressing the CANCEL key on phone triggers a Cancel notification to the application; Cisco Unified JTAPI sends CiscoCallFeatureCancelledEv to indicate the CANCEL operation. CiscoCallFeatureCancelledEv.getConsultCall() returns the earlier created consult call.

When the CANCEL operation is performed during a connected transfer or conference, the following can occur:

• The user presses the CANCEL key before selecting the Active Call softkey:

  In this case, pressing the Transfer key creates a consultCall GC3, and pressing the CANCEL key triggers CiscoCallFeatureCancelledEv on GC2 with GC3 as a consult call.

• The user presses the CANCEL key after pressing the Active Calls softkey but does this before selecting the call on phone UI.

  In this case, pressing the Active Calls softkey on the phone UI makes consultCall GC3 IDLE, but there is no CANCEL notification, as other feature operations are possible. However, if the user presses CANCEL, the CiscoCallConsultCancelEv with consult call as null, is trigerred.

• The user presses the Active Call softkey, selects a call and then presses CANCEL.

  In this case, the selected call is returned as a consultCall with CiscoCallFeatureCancelledEv.

This feature introduces interfaces that expose different configuration settings of address and terminal. These interfaces can be called even when the terminal or address is in out-of-service state. All interfaces return the values that are configured while registering with Cisco Unified Communications Manager. If the terminal is not registered, an InvalidStateException is returned. Application can get the voice mail pilot even if the terminal is not registered.

All the other changes, except voice mail, require the terminal to be reset for the new values to take effect. Interfaces return new values only after phones re-register after reset. Applications can use the interface CiscoProvTerminalRegisteredEv to read the configuration of the terminal and address.

The following configurations are exposed on CiscoAddress:

• Max calls configured

• Busy Trigger

• Position of address on a terminal

• Voice mail pilot

• ASCII and Unicode labels

CiscoTerminal provides new interfaces to applications to get the following configurations of a terminal:

• IPV4 and IPV6 IP addresses

• Outbound Rollover configuration

Terminal and address capability feature introduces new interfaces to determine if the terminal is capable of performing the following features:

• Consult call rollover

• Out bound call rollover

• Join across lines

• Direct transfer across lines

• Join on same line

• Direct transfer on same line

This enhancement restricts applications from controlling and monitoring a certain set of terminals and addresses when the administrator configures them as restricted in Cisco Unified Communications Manager Administration.

The administrator can configure a particular line on a device (address on a particular terminal) as restricted. If a terminal is added into the restricted list in Cisco Unified Communications Manager Administration, all addresses on that terminal are also marked as restricted in JTAPI. If an application comes up after the configuration is completed, it can know whether a particular terminal or address is restricted from checking the interface CiscoTerminal.isRestricted() and CiscoAddress.isRestricted(Terminal). For shared lines, applications can query the interface CiscoAddress.getRestrictedAddrTerminals(), which indicates whether an address is restricted on any terminals.

If a line (address on a terminal) is added into the restricted list after an application comes up, the applications will see CiscoAddrRestrictedEv. If the address has any observers, applications will see CiscoAddrOutOfService. When a line is removed from the restricted list, applications will see CiscoAddrActivatedEv. If an address has any observers, applications see CiscoAddrInServiceEv. Ifan application tries to add observers on an address after it is restricted, a PlatformException gets thrown. However, if any observers are added before the address is restricted, they will remain as is, but applications cannot get any events on these observers unless the address is removed from the restricted list. Applications can also choose to remove observers from an address.

If a device (terminal) is added to the restricted list after an application comes up, the application will see CiscoTermRestrictedEv. If the terminal has any observers, the application will see CiscoTermOutOfService. If a terminal is added to the restricted list, JTAPI also restricts all addresses that belong to that terminal and applications will see CiscoAddrRestrictedEv. If a terminal is removed from the restricted list, applications will see CiscoTermActivatedEv and CiscoAddrActivatedEv for the corresponding addresses. If an application tries to add observers on a terminal after it is added to the restricted list, a PlatformException is thrown. However, if observers are added before the terminal is restricted, they remain as is, but applications cannot get any events on these observers unless the terminal is removed from the restricted list.

If a shared line is added to the restricted list after an application comes up, the application will see CiscoAddrRestrictedOnTerminalEv. If any address observers exist on the address, the application will see CiscoAddrOutOfServiceEv for that terminal. If all shared lines are added to the restricted list, when the last one is added, applications will see CiscoAddrRestrictedEv. If a shared line is removed from the restricted list after the application comes up, applications will see CiscoAddrActivatedOnTerminalEv. If any observers exist on the address, the application will see CiscoAddrInServiceEv for that terminal. Ifall shared lines in the control list are removed from the restricted list, applications will see CiscoAddrActivatedEv when the last one is removed, and all addresses on terminals will receive InService events.

If all shared lines in the control list are marked as restricted, and an application tries to add observers, a platform exception is thrown. If a few shared lines are in the restricted list, while others are not, when an application adds an observer on the address. Only non-restricted lines go in service.

If any active calls are present when an address or terminal is added to the restricted list and reset, applications will see connection and TerminalConnections get disconnected.

If no addresses or terminals are added to the restricted list, this feature is backward compatible with earlier versions of JTAPI: no new events are delivered to applications.

The following sections describe the interface changes for address and terminal restrictions.

public interface CiscoRestrictedEv extends CiscoProvEv {
 public static final int ID  =  com.cisco.jtapi.CiscoEventID.CiscoRestrictedEv;

 /**
  * The following define the cause codes for restricted events
  */

public final static int CAUSE_USER_RESTRICTED  =  1;

public final static int CAUSE_UNSUPPORTED_PROTOCOL  =  2;

}

The transfer feature moves the participants of one call, the transferred call, to another call, the final call. Moving participants in a call trasitions their associated connections to the DISCONNECTED state in the transferred call and new connections for these participants getting created in the final call. Similarly, any associated TerminalConnections transition into the DROPPED state in the transferred call and get created in the final call. Cisco extensions by definition mark the start and the end of the events that relate to transfer.

You can correlate the newly created connection objects with the old connection objects by use of the CiscoConnection.getConnectionID() method to obtain the CiscoConnectionID for the old and new connections. Matching connections possess identical CiscoConnectionID objects when you compare them by using the CiscoConnectionID.equals() method.