Initializes a new instance of the VoipMediaEngine..::.ASSIGN_INCOMING_PHONE_LINE class

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.

The call ID for the call.

The host name of the call destination.

This string 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).

The destination user name of the phone call.

The INVITE request's "From:" header.

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.

The detected IP address the far end used to originate the phone call.

The detected port the far end used to originate the phone call.

This is a string that contains the received SIP INVITE message for the inbound call.

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:

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.

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 host name of the IP phone/device that initiated this call.

The sending port of the IP phone/device that initiated this call.

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

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 string contains the SIP "To:" header that is contained in received INVITE messages.