/mcp• 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. Public Interface CiscoCall getLastRedirectingPartyInfo() CiscoPartyInfo getCurrentCallingPartyInfo() CiscoPartyInfo getCurrentCalledPartyInfo() CiscoPartyInfo getCalledPartyInfo() CiscoPartyInfo Cisco Unified JTAPI Developers Guide for Cisco Unified Communications Manager, Release 15 and SUs 174 Features Supported by Cisco Unified JTAPI SIP Phone Support