Abstract:
Call screening and other communication services are described. The described methods and apparatus allow a call screening service subscriber to be provided with a caller supplied spoken name when caller ID information is blocked or unavailable. The call screening service subscriber can decide after hearing the spoken identification information how to dispose of the call. A plurality of call disposition options are supported including accept the call, reject the call, transfer to voice mail, etc. A specific salesman reject message is included as one of the call disposition options. In order to support interactive real time subscriber selection of call disposition options and the use of an answering machine by call screening service subscribers, methods and apparatus are used to detect when a human, as opposed to a machine has answered a call. When a call seeking call disposition instructions is answered by an answering machine, the caller is connected to the answering machine without further call screening being performed. Accordingly, the call screening service supports the use of home answering machines to receive messages from calls that would otherwise be blocked or disposed of if the call screening service subscriber answered his or her phone.

Description:
FIELD OF THE INVENTION 
     The present invention is directed to communications systems and, more particularly, to methods and apparatus for providing call screening and other communication services. 
     BACKGROUND OF THE INVENTION 
     In modern times, telephones have become almost a necessity. Telephones are found in most homes and offices. Marketers have found that the telephone can be used as a powerful sales tool. Telephones provide a way of reaching a potential customer who frequently would not be willing to speak with the marketer if they knew it was a salesperson calling. 
     Various attempts have been made to shield telephone subscriber&#39;s from unwanted calls, e.g., telemarketer calls. Having an unlisted telephone number provides some protection from unsolicited calls from the public at large. Computer controlled sequential dialing of multiple numbers is commonly performed by telemarketers with the express intention of reaching both listed and unlisted telephone service subscribers. Accordingly, unlisted numbers provide little protection from telemarketers. 
     Most telephone systems today use SS 7  (Signaling System  7 ) standards for communication of telephone calls. SS 7  is a digital communications protocol which supports various messaging and call information features which facilitate a variety of telephone services. SS 7  facilitates advanced intelligent network (AIN) call processing. Such processing normally includes. call handling instructions being obtained by a switch from a service control point (SCP). The SCP normally includes logic, e.g., call processing records, used to provide a switch with specific call processing instructions as a function of information obtained from a database and/or call information provided by the switch or another source. The logic in a switch used to initiate a request to an SCP for call processing instructions is normally referred to as a trigger or an AIN trigger. 
     The SS 7  messaging associated with a telephone call, includes a caller ID field which incorporates the caller&#39;s telephone number as well as a caller ID display field. By setting a caller ID blocking bit in the caller ID display field, the display of caller ID information to the called party is prohibited. Accordingly, SS 7  provides information which can be useful in identifying a caller but may be blocked from being displayed to a called party. 
     Unfortunately, in some places in this country and around the world, older analog telephone circuitry remains in use. When a telephone call is routed between telephone switches using this old analog technology, the caller ID information provided by the digital SS 7  messaging standard is normally lost. 
     In order to avoid having to answer calls from unwanted parties, e.g., telemarketers, telephone customer&#39;s often subscribe to a caller ID service or an enhanced caller ID service. With basic caller ID service, assuming the call is not passed between switches over analog lines and the caller does not activate caller ID blocking, the calling party&#39;s telephone number will be displayed to the called party. 
     In the case of enhanced caller ID service, the calling party&#39;s telephone number is used to perform a database look-up operation which associates the calling party&#39;s telephone number with a name in a database, e.g., a line information database (LIDB). Both the name and the calling party&#39;s telephone number are then displayed to the called party allowing the called party to make an educated decision as to whether or not to answer the phone call. Unfortunately, not all telephone companies share name and phone number information. In addition, even when the phone companies do exchange such information, names associated with unlisted telephone numbers may be omitted from the database used for providing name information to caller-ID service subscribers. 
     Telemarketers generally take steps to make sure that caller-ID name information is not available to local telephone companies in regions they are calling. Thus, in the case of most telemarketer calls, subscribers to caller ID services are, at best, provided a telephone number but no identifying name when being called by a telemarketer. Common caller-ID conditions which are encountered in the case of telemarketers are 1) name not available and 2) out of area. Caller ID blocked may be yet another condition which may be encountered. 
     In order to avoid disturbing telephone subscribes with calls for which caller-ID information is blocked or unavailable, a variety of call screening systems have been designed. U.S. Pat. Nos. 5,497,414 and 5,533,106 describe known call screening systems. 
     Known call screening services allow a subscriber to the service to program, prior to receipt of a call, how telephone calls for which caller-ID information is blocked or unavailable should be handled. This is done by having the service subscriber provide a list of desired call handling instructions used to create a call processing record (CPR). The call handling instructions may include, e.g., rejecting calls for which caller-ID information is unavailable or blocked, sending such calls to voice mail, or allowing calls for which a preselected call screening override code has been entered to be connected to the called party. 
     Such call screening services provide a useful tool against telemarketers and other unwanted callers. However, the known systems have several drawbacks. For example they fail to provide the call screening service subscriber the opportunity to receive calls from individuals who do not have a valid override code and whose caller-ID information is not available for legitimate reasons. For example, a calling party&#39;s caller Id information may be unavailable because the caller is traveling and calling from a pay phone or other phone for which caller-ID information is unavailable. 
     Another disadvantage of the known systems, is that calls may be blocked even when the called party is not home. From the calling party&#39;s perspective, such a situation may be undesirable since the calling party may be denied the opportunity to leave a message for the called party on an answering machine located on the called premises. From the telephone company&#39;s perspective, such a situation is undesirable since the telephone company may be denied revenue that could be collected by completing the call to an answering machine located at the called party&#39;s premises. 
     In view of the above discussion, it is apparent that there is a need for new and improved call screening methods. It is desirable that at least some of the methods provide a manner for informing a called party of a call for which caller ID information is blocked or unavailable and for allowing the called party to make a knowledgeable decision on how to dispose of the call while the calling party is still on the line. It is also desirable that at least some of the methods allow for a call to be completed to an answering machine located at the called premises or to a voicemail system. 
     SUMMARY OF THE INVENTION 
     The present invention is directed to methods and apparatus which can be used to provide call screening and other communication services. 
     In one exemplary embodiment, calls to call screening service (CSS) subscribers are detected at the central office switch to which the called party&#39;s premises are connected using a terminating attempt trigger. Upon detecting a call directed to a CSS subscriber, a check is made to see if caller ID information is blocked or unavailable. It the caller ID information is blocked or unavailable, and the calling party does not enter a call screening override code, the call is connected to an intelligent peripheral (IP) which is used to play messages to the calling and/or called party, to collect information and/or menu selection entries from the calling and/or called party, and to control ultimate disposition of the call. 
     As part of the call screening processing performed by the IP, in one exemplary embodiment, the IP records spoken caller identification information. The IP then calls the CSS subscriber to whom the call was directed. The terminating attempt trigger at the caller&#39;s switch detects the call from the IP to the CSS subscriber. However, since this second call to the caller is from the IP, the service control point responsible for providing call processing instructions to the switch result in the switch connecting the call from the IP to the subscriber&#39;s premises. 
     The IP is programmed to detect whether a human or machine answers the IP initiated call to the subscriber premises and to control subsequent call processing based on whether the call is answered by a human or machine. The manner in which a human or machine response is detected can vary depending on the embodiment. 
     In one exemplary embodiment, connection to an answering machine is determined by detecting a tone, e.g., recording prompt, or other audio or electrical signal indicative of a response from an answering machine. 
     In another exemplary embodiment, upon detecting that the call to the subscriber has been answered, the IP plays a message prompting for input from the subscriber. If the requested input, e.g., specific numbers entered using the phone key pad or spoken words, are not entered, it is assumed that a machine has answered the call and the IP connects the caller to the subscriber premises so a message may be left on the answering machine located there. Alternatively, under such circumstances, the caller may be connected to a subscriber&#39;s voicemail system. 
     If the requested input is received from the subscriber premises, it indicates that a human has answered the IP&#39;s call. Upon receiving the requested input, the subscriber is played a recording of the caller&#39;s name which was supplied by the caller and recorded by the IP. The called party is then provided a menu of call disposition options including, e.g., refuse the call, play a no salesperson message to the caller, transfer the call to voice mail (if the CSS subscriber is also a VMS subscriber) and accept the call. In response to detection of a call disposition selection made by the subscriber, the IP implements the requested disposition option. 
     In the exemplary embodiment, when the forward to voice mail option is selected by the CSS subscriber, the IP&#39;s call to the CSS subscriber is first terminated. A new call to the CSS subscriber is then initiated by the IP. This results in the terminating attempt trigger on the CSS subscriber&#39;s line to be triggered for a third time. In response to a request for call processing instructions initiated by the third trigger event, a service control point (SCP) instructs the subscriber&#39;s switch to transfer the call from the IP to the subscriber&#39;s VMS system. The calling party is then connected by the IP to the called party&#39;s VMS system where the caller can leave a message. 
     The above described call screening service makes significant use of an IP&#39;s capability to play messages, collect information, and control call disposition in response to received input. In addition, it allows a CSS subscriber to interact with the IP while a caller is on the line thereby allowing individual customized disposition of individual calls based on orally supplied caller identification information. Notably, the CSS process of the present invention is able to determine whether a call is answered by a human or an answering machine. This allows users to continue to use answering machines while still benefiting from call screening service features which allow for real time call disposition input from a called party when available. 
     Various additional features and advantages of the present invention will be apparent from the detailed description which follows. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 illustrates a communication system implemented in accordance with an exemplary embodiment of the present invention. 
     FIG. 2 illustrates an intelligent peripheral (IP) used in the system of FIG.  1 . 
     FIG. 3, which comprises the combination of FIGS. 3A and 3B, illustrates the steps of the present invention associated with processing a call directed to a call screening service subscriber. 
     FIG. 4 illustrates a voice mail service (VMS) subscriber called party selection detection routine which can be executed by the IP of FIG.  2 . 
     FIG. 5 illustrates a non-VMS subscriber called party selection detection routine which can be executed by the IP of FIG.  2 . 
     FIG. 6 illustrates steps associated with implementing a caller ID subscriber&#39;s call disposition selection in accordance with the present invention. 
     FIG. 7 illustrates the steps of an error handling routine. 
    
    
     DETAILED DESCRIPTION 
     As discussed above, the present invention is directed to methods and apparatus for providing call screening and other communication services. FIG. 1 illustrates a communication system  100  implemented in accordance with the present invention. As illustrated, the system comprises a first signal switching point (SSP)  102  which may be, e.g., a central office switch, a signal transfer point (STP)  110 , voice mail system  114 , and a second SSP  116  which are coupled to one another via a public switched telephone network (PSTN)  112 . The PSTN  112  may use, e.g., SS 7  signaling. 
     The first SSP  102  includes control logic  107  in addition to switching and input/output (I/O) interface circuitry  109 . First through third subscriber premises  117 ,  117 ′,  117 ″ are coupled to the PSTN via the first SSP&#39;s circuitry  109 . Each of the first through third subscriber premises  117 ,  117 ′,  117 ″ includes a telephone  120 ,  122 ,  124 , respectively. In addition, the first subscriber premise includes an answering machine  119  for recording messages from callers who call the first subscriber premises  117  when the subscriber is unavailable. The control logic  107  is programmed to detect calls directed to call screening service (CSS) premises which are connected to the first SSP  102 . A termination attempt trigger (TAT) is used for this purpose. The switch&#39;s control logic  107  is also programmed to seek call processing instructions, e.g., from an SCP, and thereafter follow received call processing instructions, for calls which result in a TAT being activated. 
     For purposes of explanation, assume that the telephone customer located at the first subscriber premises  117  is a call screening service subscriber. In such a case, a TAT would be set at the SSP  102  to be activated each time a call directed to the telephone number corresponding to the first subscriber premises  117  was detected at the switch  102 . 
     In order to provide AIN functionality and services such as the call screening service of the present invention, the communication system  100  includes a service control point (SCP)  106  and intelligent peripheral (IP)  104 . The SCP  106  is coupled to the SSPs  102 ,  116  via the STP  110  and PSTN  112 . The STP AND PSN are used to convey the control, data and/or voice signals as is known in the art. The IP  104  is coupled to the SCP via a TCP-IP connection  105 . This connection may be used for transferring data, e.g., call and/or input information, between the IP  104  and SCP  106 . The IP  104  is also coupled to the first switch  102 . An SS 7  communications channel  103  is used for the connection between the IP  104  and switch  102 . 
     The SCP  106  may be implemented using conventional hardware which is combined with instructions used to perform the novel call screening processing of the present invention. The SCP  106  includes call processing records, designed in accordance with the present invention, which include call handling instructions to be provided to a switch  102  in response to execution of a TAT trigger at the switch  102 . The call processing instructions associated with a particular called number vary depending on the services to which the customer, corresponding to the called number, subscribes. The instructions provided to a switch in response to a particular call can depend on: input received from the calling and/or called party, control information provided by SS 7  signaling such as ANI information, as well as other communication system status information such as the on or off-hook condition of a line at a particular point in time. The SCP  106  can access a line information database (LIDB)  108 , via STP  110 . In this manner, the SCP can obtain caller ID information, e.g., calling party name information, using a calling party&#39;s telephone number, when the information is available from the LIDB. Based on the caller ID information and status of a caller ID blocking indicator included in a call, and/or any information returned from the LIDB look-up operation, the SCP can determine whether caller ID information is unavailable, or caller ID blocked condition exists. As will be discussed below, any one of these conditions results in the SCP  106  initiating call screening procedures in accordance with the present invention. 
     The second SSP  116 , like the first SSP  102 , may be implemented using a central office switch, e.g., an SS 7  capable switch. The second SSP  116  is coupled to fourth through sixth subscriber premises  118 ,  118 ′ and  118 ″. While fifth and sixth subscriber premises  118 ′,  118 ″ are private residences which merely include telephones  126 ,  128 , respectively, the first subscriber premises  118  is a telemarketing facility. The telemarketing facility  118  includes a private branch exchange  130  and a plurality of telephones  132 ,  134 ,  136 . Using the PBX  130 , a telemarketer using one of the phones  132 ,  134 ,  136 , can sequentially call a series of telephone numbers, e.g., the telephone numbers corresponding to telephone subscriber premises  117 ,  117 ′,  117 ″. Assuming that the telephone subscriber located at the first subscriber premises  117  subscribes to the call screening service of the present invention, a telemarketing call directed to the premises  117  would result in a TAT being executed at the first switch  102 . 
     The intelligent peripheral (IP)  104  is illustrated in greater detail in FIG.  2 . As illustrated the IP  104  includes a DTMF detector/generator circuit  202 , a text to speech (TTS) circuit  204 , a speech recognizer  206 , audio recording and playback circuitry  208 , a central processing unit CPU  212 , memory  213  and switching and I/O circuitry  224  which are coupled together by a bus  210 . The switching and I/O interface circuitry  224  is coupled to the SSP  102  via communications line  103  and to the SCP  106  via TCP/IP connection  105 . The circuitry  224  is responsible for performing switching operations and for converting between protocols used on the communication lines  103 ,  105  and various components coupled to the internal bus  210  thereby allowing the exchange of instructions, data and other signals between the SSP  102 , SCP  106  and the various components of the IP  104 . 
     The DTMF detector/generator  202  is used for detecting DTMF input from a caller and for generating DTMF signals used to place a call through the switch  102 . TTS circuit  204  is capable of generating audible speech from electronic text prompts. The TTS circuit  204  is useful for prompting a caller for input and/or for playing messages to a party to thereby provide the party with call or service related information. The speech recognizer  206  is capable of recognizing speech. In various embodiments, it is used to detect spoken digits received in response to a request for a numerical input, e.g., a numbered menu selection. The audio recording/playback circuit  208  provides speech recording and playback capability. In various embodiments, it is used to store verbal identification information, e.g., a spoken name, obtained from a calling party and to later playback the recorded information to a called party. 
     The CPU  212  is responsible for controlling IP operation under direction of instructions included in the various routines stored in the memory  213 . As illustrated, the memory  213  includes call screening service subscriber information  214 , a set of text prompts  216 , CSS control routines  218  and audio recordings  222 . In another embodiment, CSS information  214  is stored in the SCP  106  as opposed to the IP  104 . 
     The CSS subscriber information  214  is stored in the IP and/or SCP, includes lists of CSS subscribers, is identified by their corresponding telephone numbers, information on whether they are also voice mail service (VMS) subscriber&#39;s, one or more call screening override codes and related service billing information. As will be discussed below, the CSS subscriber information  214  is accessed and used by the CSS control routines  218  in controlling operation of the IP  104  to service a call directed to a call screening service subscriber. Individual prompts included in the set of prompts  216 , are supplied to the speech generator  204  as required when performing a call screening operation. Audio recordings  222  include recordings of spoken identification information, e.g., caller&#39;s names, generated by recording circuit  208 . As will be discussed below, the recording of a calling party&#39;s speech, e.g., spoken name, is played to a called party at specific times while performing call screening in accordance with the present invention. 
     The CSS control routines  218  are executed by the IP  104  when a call screening service is to be performed. The steps performed by the IP under direction of the CSS control routines  218  will be discussed in detail below with regard to FIG.  3 . 
     FIG. 3, which comprises the combination of FIGS. 3A and 3B, illustrates the call processing method  300  of the present invention. The method begins in start step  302  wherein the components of the system  100  are initialized. For example, in step  302  an AIN terminating attempt trigger (TAT) is set at the switch  102  on each of the lines corresponding to a call screening service subscriber. For purposes of explanation, it will be assumed that the telephone customer located at customer premises  117  is a call screening service subscriber. In such a case, in step  302 , a TAT trigger is set to detect calls received at the switch  102  that are directed to the telephone number corresponding to subscriber premises  117 . 
     Once the triggers are set in step  302  operation proceeds to step  304 . In step  304  the switch  102  is operated to use the triggers to detect calls directed to call screening service subscribers. 
     Upon detecting a call to a call screening service subscriber, e.g., a call directed to customer premises  117 , the TAT set at switch  102  is activated and operation proceeds to step  306 . In step  306 , in response to a call to customer premises  117 , the switch  102  initiates a call processing instruction request to the SCP  106 . As part of the request, the switch  102  passes called party identification information, e.g., the telephone number called, calling party identification information, e.g., ANI information, and caller-ID blocking status bit information to the SCP  106 . 
     In response to the first request for call processing instructions, in step  308 , the SCP  106  determines if the caller ID information is blocked or unavailable. This is done by examining the contents of the Calling Party ID parameter in the call processing query message sent to the SCP. If the calling party number is blank or the caller-Id blocking bit, which may be set by the caller, is set to prohibit display of caller ID information, the SCP  106  concludes that the caller ID is unavailable or blocked. 
     In step  308 , if it is determined that the caller ID information is not blocked and is available, the SCP returns the caller ID information to the switch  102  and instructs the switch to allow the call to be completed to the called CSS subscriber  117 . In step  310  the switch  102  is operated to complete the call to the called CSS party, e.g., subscriber premises  117 , and the call is then allowed to terminate in a normal manner, e.g., with one of the parties hanging up. 
     However, if in step  308 , it is determined that the calling party has the caller ID blocked or that caller ID information is unavailable, the SCP  106  instructs the switch  102  to use the IP  104  to obtain additional information, e.g., identification information from the calling party, and operation proceeds to step  312 . The switch  102  does this, in one embodiment, by performing a send to outside resource operation in response to the instructions from the SCP  106  where the outside resource is an IP  104 . 
     In step  312 , a retry counter, RC, which may be maintained by the IP  104 , is initialized to 0. Next, in step  314 , the IP  104  is used to play a message, e.g., one of the prompts  216 , to the calling party using speech generator  204 . The message states: “THE CALLED PARTY HAS CALL SCREENING AND DOES NOT ACCEPT CALLS FROM UNIDENTIFIED NUMBERS”. Then, in step  316  the IP  104  is used to play another message to the calling party. This time the message states: “TO RECORD YOUR NAME, PLEASE PRESS THE # KEY OR SIMPLY STAY ON THE LINE”. 
     In response to this message the caller can, optionally, enter a call screening override code. In this manner, a family member or other individual to whom the called party has provided override code information can override the call screening process and be connected to the called party even when caller ID information is blocked or unavailable. 
     From step  316 , operation proceeds to step  318  wherein the SCP  106  detects entry of an override code, entry of the pound symbol (#), or the occurrence of a timeout condition. In step  320 , a determination is made as to whether or not an override code was entered. If an override code, e.g., one or more DTMF signals other than the # symbol, was entered operation proceeds to step  322 . 
     In step  322  a check is made to determine if the override code was valid. This may involve a comparison of a received override coded to one or more valid override codes stored in the CSS subscriber information  106  for the CSS subscriber to whom the call was directed. If the received override code is valid for the called party, operation proceeds to step  310  wherein the switch  102  connects the calling party to the called party. 
     However, if the override code is determined in step  322  to be invalid, operation proceeds to step  324  wherein the retry counter RC is incremented by one. Then in step  325  the value RC is compared to 4. If RC is less than 4, operation proceeds to step  327  in order to provide the calling party another opportunity to enter an override code or provide name information. In step  327 , the IP plays the calling party a message stating: “I&#39;M SORRY I DID NOT UNDERSTAND WHAT YOU PRESSED. PLEASE TRY AGAIN”. With the playing of the message, operation proceeds from step  327  to step  316 . 
     RC equaling or exceeding  4  indicates that the calling party has already had three unsuccessful attempts at entering an override code. If in step  325  it is determined that RC is not less than 4, operation proceeds from step  325  to step  326 . In step  326 , the caller is played a message stating: “THERE IS AN INPUT ERROR. GOOD BYE.” Then, in step  328 , the call is terminated by the switch  102 . 
     In step  320 , if it is determined that an override code has not been entered, operation proceeds to step  330 . In step  330 , the switch  102  is instructed to disconnect from the IP  104  . Then, in step  332 , the switch  102  is controlled to forward the call being processed to the IP  104 . At this point, the transaction between the switch  102  and the SCP  106  which was initiated in response to the first call to the CSS subscriber  117  is closed and the IP  104 , under control of the CSS control routines  218 , takes over call processing. Then in step  333  the retry counter RC is reset to 0. From step  333  operation proceeds to step  334 . 
     In step  334 , the IP  104  plays a recording prompt to the caller stating: “AT THE TONE, PLEASE SAY YOUR NAME OR THE COMPANY YOU REPRESENT, THEN PRESS THE POUND KEY”. Then, in step  336 , the IP records the audio from the caller until a time out condition occurs or entry of a # signal is detected, e.g., by the DTMF detector  202 . 
     In step  338  a determination is made as to whether or not speech, e.g., a name, has been recorded. This step may be made by distinguishing from a recording of silence as opposed to speech. If any speech was recorded, it is assumed to be a name since a name was requested. Any one of a plurality of known techniques may be used to implement step  338 . 
     If in step  338  if it is determined that speech has not been recorded, operation proceeds to step  340  wherein the retry counter RC is incremented. Operation then proceeds to step  342 . 
     In step  342 , RC is compared to  4 . If RC&lt; 4 , then operation proceeds to step  344  to provide the caller another opportunity to record a name. In step  344  the IP  104  plays a message to the caller stating: “THE NUMBER YOU ARE CALLING HAS CALL INTERCEPT AND DOES NOT ACCEPT CALLS FROM UNIDENTIFIED NUMBERS”. Operation then proceeds once again to step  334 , wherein the caller is prompted to provide a name. 
     If in step  342  it is determined that RC is not less than 4, i.e., the caller has already been provided three chances to leave a name, operation proceeds to step  346 . In step  346 , the caller is played a message stating: “YOU HAVE NOT RECORDED YOUR NAME. THE PERSON YOU ARE CALLING DOES NOT ACCEPT CALLS FROM UNKNOWN OR BLOCKED NUMBERS. GOOD BYE”. Then, in step  348 , the call is terminated with the calling party being disconnected. 
     In step  338 , if it is determined that a name provided by the calling party was recorded, operation proceeds to step  350  wherein the IP  104  places a call to CSS subscriber  117  via switch  102 . This causes the TAT trigger on the CSS subscriber&#39;s line to be activated a second time launching a second request to the SCP  106  for call processing instructions. Recognizing the IP  104  as the calling party, the SCP instructs the switch  102  to complete the call from the IP to the CSS subscriber  117 . 
     Via connection nodes  352  and  354  which serve to link FIGS. 3A and 3B together, operation proceeds from step  350  to step  356 . In step  356 , the IP  104  plays music to the waiting caller. Then in step  358  the IP  104  is operated to monitor for an answer from the subscriber located at the called premises  117  or for the occurrence of a time out condition. An answer may be detected by examining the hook-status of the called party&#39;s line. The occurrence of an off-hook condition, in response to the IP&#39;s call to the CSS subscriber, indicates an answer. 
     If no answer is detected in step  360 , operation proceeds to step  362  wherein the IP  104  plays a message to the calling party stating: “THE CALLED PARTY IS UNAVAILABLE”. Then in step  364  the call is terminated with the calling party being disconnected. Alternatively, if the called party is a VMS subscriber, the call may be completed to the subscriber&#39;s VMS system. 
     If an answer is detected in step  360 , operation proceeds to step  368  after the retry counter RC is reset to 0 in step  366 . In step  368  the IP  104  plays a message to the called party stating: “SOMEONE IS WAITING TO SPEAK WITH YOU. FOR MORE INFORMATION, PRESS ONE.” Then in step  370  DTMF input or the occurrence of a time out condition is detected. In step  372 , a determination is made as to whether or not DTMF input was detected in step  370 . 
     If it is determined in step  372  that DTMF input was not received, it is assumed that an answering machine has answered the call to the CSS subscriber&#39;s premises  117 , and operation proceeds to step  376 . Block  374  which states the assumption being made is not an actual processing step but is included for purposes of explanation. In step  376 , the calling party is connected to the called CSS subscriber premises  117  thereby allowing the calling party to leave a message on the answering machine  119 . In step  377 , the call is allowed to terminate in a normal manner, e.g., with either the calling party or the answering machine  119  terminating the call by hanging up. 
     In step  372 , if it is determined that a DTMF input was received, operation proceeds to step  378  wherein a determination is made as to whether or not the requested number “one” was received in DTMF format. If it is determined that a one was not received, operation proceeds to input error handling subroutine  700  wherein the IP seeks additional input or terminates the call after a preselected number of tries. The input error handling routine  700  will be described below in detail with regard to FIG.  7 . Upon returning from the error handling sub-routine  700  operation proceeds to step  370  wherein input from the called party or the occurrence of a time out condition is once again detected. 
     If in step  378 , it is determined that a one was received, indicating that a human operator provided a response to the message about a waiting caller, operation proceeds to step  382 . In step  382 , the IP plays the subscriber a message stating: “CALL FROM:”. Then in step  384 , the IP  104  plays to the called party, the recorded audio of the calling party&#39;s speech which was obtained in response to a request for a name. Next in step  386 , the retry counter RC is reset to 0. Then in step  388  a determination is made as to whether or not the called CSS subscriber is also a voice mail service (VMS) subscriber. The menus of call disposition options provided to the called party vary depending on whether or not the called party is a VMS subscriber. 
     If the called party is a VMS subscriber operation proceeds to the called party selection detection routine  400  via step  390 . However, if in step  388  it is determined that the called party is not a VMS subscriber, operation proceeds to the non-VMS subscriber called party selection detection routine  500 . 
     The VMS subscriber called party selection detection routine  400  begins in start step  402  of FIG. 4 wherein it begins being performed by the IP  104  under control of CPU  212 . Operation proceeds from start step  402  to menu step  404 , wherein the IP plays a menu to the called party. In one exemplary embodiment, it does this by playing the message: “TO ACCEPT THIS CALL, PRESS 1; TO DENY THIS CALL, PRESS 2; TO PLAY THE SALES CALL REFUSAL TO THE CALLER, PRESS 3; TO SEND THIS CALL TO VOICE MAIL, PRESS 4; TO REPLAY THE CALLERS NAME, PRESS 5”. 
     Next, in step  405  user input or the occurrence of a time out condition is detected by the IP  104 . Then in step  406  a determination is made as to whether or not a valid DTMF input was received from the called party. That is, a determination is made as to whether a number on the menu played in step  404  was received. If a valid input was received by the IP  104  from the called party, operation proceeds to the selection implementation sub-routine  600  via step  410 . However, if a valid input was not received from the called party, operation proceeds to step  408  wherein the input error handling sub-routine  700  is called. Upon returning from the input error handling sub-routine, operation proceeds from step  408  to step  404  wherein the menu of available call disposition options is again played to the called party. 
     The NON-VMS subscriber called party selection detection routine  500  begins in start step  502  of FIG. 5 wherein it begins being performed by the IP  104  under control of CPU  212 . Operation proceeds from start step  502  to menu step  504 , wherein the IP plays a menu to the called party. In one exemplary embodiment, it does this by playing the message: “TO ACCEPT THIS CALL, PRESS 1; TO DENY THIS CALL, PRESS 2; TO PLAY THE SALES CALL REFUSAL TO THE CALLER, PRESS 3; TO REPLAY THE CALLERS NAME, PRESS 5”. Note that this menu is the same as that provided in step  404  to the VMS subscriber with the exception that the voice mail option is not presented to the called party since the party does not subscribe to the VMS service. 
     Next, in step  505  user input or the occurrence of a time out condition is detected. Then in step  506  a determination is made as to whether or not a valid DTMF input was received from the called party. That is, a determination is made as to whether a number on the menu played in step  504  was received. If a valid input was received by the IP  104  from the called party, operation proceeds to the selection implementation sub-routine  600  via step  510 . However, if a valid input was not received from the called party, operation proceeds to step  508  wherein the input error handling sub-routine  700  is called. Upon returning from the input error handling sub-routine  700 , operation proceeds from step  508  to step  504  wherein the menu of available call disposition options is again played to the called party. 
     FIG. 6 illustrates the selection implementation sub-routine  600 . The routine  600  begins in step  602  and proceeds to step  604  wherein the processing path to be followed is determined as a function of the DTMF input, e.g., value, received from the called party. Step  604  may be implemented using a case statement as is known in the programming art. 
     If a  1  is received as the menu selection from the called party, path  1  is followed from step  604  to step  606 . In step  606  the IP  104  plays a message to the called party stating:. “NOW CONNECTING”. Then in step  610 , a determination is made as to whether or not the calling party is still connected to the switch, i.e., the calling party has not hung up while waiting for the called party. 
     If in step  610  it is determined that the calling party is still connected operation proceeds to step  612  wherein the calling and called parties are connected together. After the calling and called parties are connected by the IP  104 , the call is allowed to terminated in step  614  in a normal fashion, e.g., with one of the parties hanging up. 
     If, however, in step  610  it is determined that the calling party is no longer connected, e.g., because they hung up, the IP  104 , in step  616 , plays the calling party a message stating “WE&#39;RE SORRY. THE PERSON WAITING TO SPEAK WITH YOU HAS HUNG UP”. The call is then terminated in step  618 . 
     If a  2  is received as the menu selection from the called party, path  2  is followed from step  604  to step  620 . In step  620  the IP  104  plays a message to the called party stating: “CALL DENIED”. This announcement is followed in step  622  with the termination of the connection between the IP and the called party. In step  624  the calling party is played a message “THE PERSON YOU ARE CALLING IS NOT AVAILABLE. THANK YOU. GOOD BYE.” The call is then terminated in step  618  with the calling party being disconnected from the IP  104  and switch  102 . 
     If a  3  is received as the menu selection from the called party, path  3  is followed from step  604  to step  630 . In step  630  the IP  104  plays a message to the called party stating: “THE SALES CALL REFUSAL MESSAGE WILL BE PLAYED TO THE CALLER.” This announcement is followed in step  632  with a message being played to the calling party. The message played to the calling party states: “THE PERSON YOU ARE CALLING DOES NOT ACCEPT PHONE SOLICITATIONS. PLEASE ADD THEIR NAME TO YOUR DO NOT CALL LIST. THANK YOU. GOOD BYE.” The call is then terminated in step  618 . 
     If a  4  is received as the menu selection from the called party, path  4  is followed from step  604  to step  640 . In step  640  the IP  104  plays the message “THE CALLER HAS BEEN SENT TO VOICE MAIL” to the called party. This announcement is followed in step  642  with the termination of the connection between the IP and the called party. In step  644  the calling party is played a message “NOW CONNECTING TO AN ANSWERING SYSTEM.” In step  646 , the IP initiates a new call to the CSS subscriber&#39;s premises  117 . This call causes the TAT on the subscriber&#39;s line to be activated for the third time. In response to activation of the TAT the switch  102  initiates a new inquiry to the SCP  106  for call processing instructions. At this point in time, the calling party is still connected to the IP  104  and the SCP  106 . The SCP  106  detects from the call information provided to it that this is the second call from the IP to the CSS subscriber in regard to the call from the calling party. In response to this-second call from the IP, the SCP instructs the switch  102  to connect the call to the CSS subscriber&#39;s VMS  114 . The IP  104 , in step  648 , connects the calling party to the called party&#39;s VMS. Then in step  650  the calling party is provided an opportunity to leave a message for the called party prior to the call being terminated in step  618 . 
     If a  5  is received as the menu selection from the called party, path  5  is followed from step  604  to step  650 . Step  650  is a GOTO STEP. In step  650 , operation proceeds to step  384  and then to step  404  if the called party is a VMS subscriber and to step  504  if the called party does not subscribe to voice mail. Thus, via the path provided by step  650 , the called party is provided an opportunity to hear the menu of available call disposition options again. 
     An exemplary input error handling sub-routine  700  which may be used by various other IP control routines and steps, is illustrated in FIG.  7 . The routine  700  is used to determine if the party providing input should be provided another opportunity to input the request information or menu selection or the call should be terminated. In the FIG. 7 embodiment, a party is given a total of 3 chances to input expected data with the retry counter RC being used to determine when the three chances have been provided. 
     Operation proceeds from start step  702  to step  704  wherein the retry counter RC is incremented by one. Then, in step  706  a determination is made as to whether RC is less than 4. If RC is not less than 4, e.g., 4 or greater, three chances have already been provided to supply the expected input and operation proceeds to step  708 . In step  708 , the IP plays a message to the called party stating: “I&#39;M UNABLE TO UNDERSTAND YOUR RESPONSE. GOOD BYE.” Then in step  710  the call is terminated. 
     However, if in step  706  it is determined that RC is less than 4, operation proceeds to step  712  wherein the IP plays the message: “I&#39;M SORRY I DID NOT UNDERSTAND WHAT YOU PRESSED. PLEASE TRY AGAIN.” Operation then returns in step  714  to the routine or sub-routine which called the input error handling routine  700  to allow another chance to enter the expected input. 
     Through the above discussed process, a subscriber can be shielded from calls with blocked or unavailable caller ID information while allowing the CSS subscriber to specify call disposition options in real time. In addition, because the process provides for handling responses from a caller&#39;s answering machine, the processes of the present invention is compatible with the use of home answering machines. 
     In the embodiment described above responses to prompts other than a name prompt are normally entered by depressing telephone keys. However, the system of the present invention can, and in one embodiment does, use speech recognition techniques to allow a CSS subscriber to enter responses using speech. In such an embodiment, a CSS subscriber may state “one” to select call disposition option one from the menu of call disposition options or enter “1” using a telephone keypad. A spoken “1” detected by the IP&#39;s speech recognizer  206  while a “1” entered using the telephone keypad is detected by DTMF detector/generator circuit  202 . 
     While the detection of calls directed to a CSS subscriber has been described as being performed at the switch  102  to which the lines to the subscriber premises are connected, it is to be understood that the same functionality may be implemented elsewhere in the system  100 , e.g., at another switch through which calls are routed, using a similar trigger to detect calls to CSS subscriber&#39;s. 
     Numerous additional embodiments, within the scope of the present invention, will be apparent to those of ordinary skill in the art in view of the description included herein and the claims which follow.