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Method and apparatus for managing telephone calls

Method and apparatus for managing telephone calls description/claims

The present invention relates generally to call processing and, more particularly, to a system, method, and computer program product for processing and managing communications over network based servers operating over IP protocols.

Voice telephony and communications today is generally provided to subscribers via the public switched telephone network (PSTN), wireless communications networks, and the Internet, which can carry telephone calls between PSTN gateways, from end user to end user, or between an end user on the Internet and an end user on the PSTN or wireless network.

The process of setting up telephone calls for the PSTN is well known and has used “out of band” communication over the SS7 (“Signaling System 7”). Generally, SS7 signaling uses a network and protocol that is separate from the network over which voice traffic flows and is used to control switches, such as an AT&T ESS #4 switch, in circuit switched networks. Additional protocols have been used for the routing of voice data over the Internet, including International Telecommunications Union H.323 Internet protocol and the Session Initiation Protocol (SIP).

Call processing features, such as call forwarding, have been made available to PSTN subscribers via SS7 control protocols, and separately to wireless subscribers and VOIP providers. In addition, the PSTN has been used to provide one number telephone routing, pursuant to which a single telephone number is used to reach a subscriber at one of several telephones. Toll free calling has also incorporated call routing to multiple telephone number based on time of day and other features.

Systems in the prior art have attempted to provide call processing features. Each of these prior systems, however, have many disadvantages and/or deficiencies. Generally, the prior art does not generally allow customization down to the individual caller level. For instance, a system provided by Ring Central, found at www.ringcentral.com, provides call processing features. Ring Central, however, does not allow settings for each web call to be customized. Moreover, ring back tones cannot be uploaded for use with their system. Importantly, calls cannot be recorded, either in their entirety or at a time selected by the user. The prior art systems also fail to remember the identify of a caller who has previously called. Thus, a caller is forced to identify themselves each time they call, which causes a significant inconvenience.

Moreover, Ring Central and other prior art systems do not allow a call to be switched to other phones designated by a user. Though the systems often provide voicemail capabilities, they do not allow a user to eavesdrop on a voicemail message while it is being left, or to join in with the caller leaving a voicemail. In addition, prior art systems fail to allow text to speech screening.

However, there remains a need for systems and methods to provide a comprehensive communications solution to a subscriber to enable the subscriber to manage communications via separate subscriber accounts through a single portal and to take advantage of data networks and real time protocols using data networks to the maximum extent possible in managing communications. There is still a further need for systems and methods to interface with the PSTN and other networks through gateways that allow management of control of routing through those networks for the convenience of subscribers with accounts on one or more of such networks.

According to the present invention, a subscriber uses a single telephone number and servers accessible through an Internet portal to manage the routing of communications among separate communications networks. This is accomplished through a communications system that exchanges communications, such as calls, pursuant to real time protocols with other networks, including the PSTN, wireless networks and VOIP networks, and that exchanges control signals with those same networks through a control protocol, such as the session initiation protocol (SIP). The communications systems and methods described herein allow the routing of inbound calls to a single telephone number outbound to one or more of the following: a voicemail database; a PSTN telephone number; multiple PSTN telephone numbers; wireless telephone numbers; VOIP telephone numbers; and other addresses. In addition, a caller may initiate an outbound call from the system via the Internet portal to connect one of the subscriber's telephones to a destination telephone.

According to one embodiment of the present invention, a system for processing calls includes: a database including configuration information and authentication information for users; an inbound controller coupled to an inbound gateway and the database that exchanges first control signals over a session initiation protocol (SIP) with a communications gateway that generates second SIP control signals based on information in the first control signals and the configuration and authentication information; a voicemail database that stores and retrieves communications; a routing controller that prompts callers for information and instructions; an outbound communications controller coupled to an outbound gateway that exchanges fourth control signals with the outbound gateway and communications via a real time protocol with the outbound gateway in order to send the inbound communications outbound over at least one communications network. The system further includes a switch coupled to the inbound controller, the inbound and outbound communications gateways, the voicemail database and the routing controller. The switch may receive the inbound communications over a real time protocol from the inbound gateway, exchange the second control signals with the inbound controller, and route inbound communications based on the second control signals and the routing controller to at least one of the following destinations: the voicemail database; and at least one communications network by sending outbound communications via a real time protocol and exchanging third control signals with the outbound controller, including routing information relating to the outbound communications.

According to various embodiments, the switch may route the inbound communications to the voicemail database and to at least one telephone on at least one of the communication networks. In addition, the routing controller may include a state machine coupled with a script database, which stores a configurable script used to issue voice prompts to a caller at one end of one of the inbound communications system for information pertaining to the handing of that inbound communication.

According to another embodiment, the present invention comprises a method for processing calls. The method includes operatively connecting a switch to an inbound controller, an inbound gateway, an outbound controller, and an outbound gateway. First control signals may then be exchanged over a session initiation protocol (SIP) between the inbound gateway and the inbound controller based on inbound communications to generate second SIP control signals based on information in the first control signals and configuration and authentication information. The switch preferably receives inbound communications over a real time protocol from the inbound gateway, and the second control signals are exchanged with the inbound controller. Fourth forth control signals are also exchanged between the outbound controller and the outbound gateway via a real time protocol in order to send the inbound communications outbound over at least one communications network. Finally, inbound communications are routed based on the second control signals and the routing controller to at least one of: (i) a voicemail database; or (ii) at least one communications network by sending outbound communications based on the inbound communications to the outbound controller via a real time protocol and exchanging third control signals with the outbound controller, including routing information relating to the outbound communications.

According to another aspect, the present invention comprises a method for processing calls. The method preferably includes receiving a call from a caller at an origination point on a first network, and then exchanging communications signals between the first network and a second network using a session initiation protocol. Preferably, the communications signals at the second network are processed using configuration and authentication information. The method also includes determining at least one destination point from a group of destination points based on the processing and directing the communications signals to the at the at least one destination point and, prior to connecting the caller and the user, notifying a user of the call from the origination point.

In one embodiment, the notifying step comprises allowing the user to select from the following options: (i) accept the call; (ii) send the call to a voicemail database; (iii) accept the call and record the call in a database; or (iv) send the call to the voicemail database and simultaneously listen to the call. It may be desirable to allow the call to be directed to two or more destination points substantially simultaneously. Optionally, an audible response is sent to the origination point based on the configuration information and authentication information. The audible response includes at least one of: a song (or other MP3); and a voice recording. It may be desirable for the group of destination points to comprise a group of telephones.

A user may accept the call on at least one telephone after being notified, and preferably the user has the option to notify each of the telephones within the group of telephones of the call. The call may be transferred from a first telephone to a second telephone when the second telephone indicates acceptance of the call. Alternately, the at least one destination point comprises a voicemail database. The voicemail database preferably stores audio data, and is operable to convert the audio data into text data. The text data stored in the voicemail database is searchable.

According to another embodiment, a graphical representation of data in the voicemail database is displayed on a webpage. The graphical representation includes at least some of: the name of a caller; the time of the call; the date of the call; a duration of the voicemail; and the location of the caller. Preferably, the voicemail database is arranged based on a predetermined order selected by the user. To allow increased accessibility, the database may be accessible from a remote location. Optionally, a user can select a link on the webpage that connects a telephone selected by the user to the caller. The user can accept the call and selectively record at least a portion of the call.

According to another aspect, the present invention comprises another method for processing calls. The method includes receiving a call from a caller at an origination point on a first network, and then exchanging communications signals between the first network and a second network using a session initiation protocol. The communications signals at the second network are preferably processed using configuration and authentication information. Two or more destination points from a group of destination points are also determined based on the processing. Then, a signal is sent to each of the two or more destination points substantially simultaneously. A user input determines which of the destination points a communication is established with. A user at the one selected destination point is then presented with identifying information about the caller and a plurality of options for handling the call. If the user chooses to accept the call, the user is allowed to transfer the call by providing an input that re-initiates the sending step.

In one embodiment, the present invention queries a first time caller for their identity. Thereafter, the present invention is operable to store the identity of the caller in a memory. This feature prevents the caller from having to identify themselves each time they call. Instead, the callers identity, e.g., their name, can be provided to the user. If a first time caller's number is already in the user's address book, the present invention will not ask the caller for their identity. Instead, the present invention will read the caller's name from the user's address book, e.g., using a text to speech feature. Preferably, the caller preferably hears an audio file pre-selected by the user after the processing step. The plurality of options for handling the call include at least one of: sending the call to a voicemail database and listening in on the voicemail as it is being recorded; or accepting the call and simultaneously recording the call. If the user selects the option of listening in on the voicemail, the user can selectively terminate the voicemail and begin communicating with the caller by providing an input. Alternately, the user can send the call to a voicemail database, and the contents of the voicemail database, e.g., duration of a call, name of the caller, telephone number of the caller, are displayed on a webpage. It may be desirable for the order of the voicemails displayed on the webpage to be customized by the user. If, however, the user chooses to accept the call after the presenting step, the user can selectively record at least a portion of the call into a database.

According to yet another aspect, the present invention comprises a computer program product for use in a call processing system. The system includes a computer readable medium and computer program instructions, recorded on the computer readable medium, executable by a processor, for implementing a call processing system. The call processing system includes the steps of receiving a call from a caller at an origination point on a first network and exchanging communications signals between the first network and a second network using a session initiation protocol, and processing the communications signals at the second network using configuration and authentication information.

Two or more destination points from a group of destination points are then determined based on the processing, and then a signal is sent to each of the two or more destination points substantially simultaneously. A communication is then established with one selected destination point based on a user input, and a user at the one selected destination point is presented with identifying information about the caller and a plurality of options for handling the call. If the user chooses to accept the call, the user can transfer the call by providing an input that re-initiates the sending step.

• Provisional Patent
• Utility Patent

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