ASSIGN_INCOMING_PHONE

The ASSIGN_INCOMING_PHONE_LINE structure is sent to an application when the media engine receives an incoming call. The media engine uses it to convey information to the application regarding the incoming call and to allow the application a chance to determine what available input phone line the call will be assigned to. Also, the application can set a member variable of this structure to allow the media engine to immediately terminate the incoming call. Application software can access the information in this structure when it processes the SipIncomingCallAssignPhoneLine event.

typedef struct

{

char *pReceivedSipInvite;

// Source information

char *pFromHeader;

char *pSrcUrl;

char *pSrcDisplayName;

char *pSrcUserName;

char *pSrcContactUserName;

char *pSrcHost;

DWORD SrcPort;

char *pReceivedIpAddress;

DWORD ReceivedPort;

// Destination information

char *pToHeader;

char *pDestUrl;

char *pDestUserName;

char *pDestHost;

char *pCallId;

// Other information

char *pOriginatingEngineId;

// data returned to the media engine.

int AssignedPhoneLine;

BOOL TerminateIncomingCall;

int SipResponseCode;

}ASSIGN_INCOMING_PHONE_LINE;

This is a pointer to a NULL terminated string that contains the received SIP INVITE message for the inbound call.

This character array allows application software to access the SIP "From:" header that is contained in received INVITE messages.

This character array contains the SIP URL of the person calling us. Please refer to the installed SIP RFCs for a description of the format of SIP URLs (RFC 3261). The information contained in this URL is equivalent to the information that is present in the received INVITE message "From:" header.

The display name of the source of the phone call. This string may be empty if the far end that is initiating the call does not have a display name assigned.

The name of the IP phone/device that initiated this call. This is the same as the user name specified in the SIP "From:" header.

The name of the IP phone/device that initiated this call. This is the same as the user name specified in the SIP "Contact:" header.

This character array allows application software to access the SIP "To:" header that is contained in received INVITE messages.

This character array contains the SIP URL for the destination of the phone call. If you are developing a PSTN gateway, you will use this member to get the PSTN phone number. Most applications can ignore this member. Please refer to the installed SIP RFCs for a description of the format of SIP URLs (RFC 3261).

If the originator of the phone call is another LanScape VOIP Media Engine, this is the unique ID of that media engine. Application software can ignore this ID value.

AssignedPhoneLine
This value is set to -1 by default by the media engine. Application software can set this value to a zero based phone line index that represents the phone line that should be used to answer the incoming phone call. If application software does not want to tell the media engine what phone line to assign the incoming call to, then this value should not be modified.

TerminateIncomingCall

Application software can set this value to a non zero value to allow the VOIP Media Engine to terminate the incoming call immediately. If your application wants the VOIP Media Engine to process the received call as normal, it should set this value to zero (FALSE).

This is the SIP response code you want your application to return to the far end of the incoming call. Used only when terminating the call. Any one of the following values can be specified:

If you specify any other 4xx value that is not described above, the VOIP Media Engine will ignore your value and return 480 to the far end of the call.