MXPA00002891A - Method and apparatus for dynamically routing calls in an intelligent network - Google Patents

Abstract

A method, apparatus and system for dynamically routing selected calls through an intelligent switched telephone network are described. The method leverages the resident switching power in the Public Switched Telephone Network (202) by departing from the Advanced Intelligent Network (AIN) call model while adhering to the basic principles of ISUP common channel signaling to introduce new flexibility in call routing. Using the method, calls can be efficiently routed and rerouted through the network. Control of a call can be effected by either the called party or the calling party. The method can be practised using either a virtual switching point (VSP) (208) or an ISTP (232). The VSP is a physical node in the signaling plane of the network and a virtual node in the switching plane. Calls are routed to the VSP using dedicated trunk groups which may be loop-back ISUP trunks (234) or inter-switch ISUP trunks (212). Calls are routed to the dedicated trunk groups using standard routing translation tables and methods. The advantage is a new level of flexibility in call routing control that permits the rapid introduction of new services which include features that could not be efficiently accommodated using prior methods of call routing.

Description

METHOD AND APPARATUS FOR DIRECTING DYNAMICALLY CALLS IN AN INTELLIGENT NETWORK TECHNICAL FIELD This invention relates to the direction of calls through an intelligent switched telephone network and, in particular, to a novel method and apparatus for dynamically redirecting calls through the network, without the disconnection of the calling party, in response to any predefined criteria, which uses the standard signaling messages of the common channel in a call control node, which is a virtual node in the switching plane and a physical node in the signaling plane of the network.

BACKGROUND OF THE INVENTION The use of the telephone as a social and business instrument has grown exponentially in the past 100 years. The wide acceptance of the telephone and its use has stimulated the industry to create many innovations to facilitate the consummation of calls and increase telephone services. The current communication of telephone users is sophisticated in the use of telecommunications equipment and demands for faster connections, greater services and better integration with computer applications to help in modernizing their business operations. The Telephone Network Switched to the Public (PSTN) has developed a computer-controlled, highly automatic switched network that allows callers to place calls virtually anywhere in the world. In this document, the use of the term PSTN is intended to refer to any intelligent switched telephone network. Many of the PSTNs are now referred to as the Intelligent Advanced Network (AIN). In the AIN, the "triggers" embedded in the factory of the network switching nodes, allow the call requests to initiate queries to the database to search the call address information. After the call address information is returned from a consulted database, the call connects through the network using standard procedures. Although the AIN is a high speed, multi-aspect network, which provides a vast array of automatic telephone services, the service development in the AIN is channeled through the AIN call model. In the AIN, the new service development is achieved using Service Creation Environments to create logical service programs, which are executed by the Intelligent Service Control Points (ISCP), which are databases that respond to switching queries. initiated by the activators of the AIN. In the AIN call model, the opportunities to initiate management decisions are essentially limited to the activation points embedded in the network switching factory. Although the services offered in the PSTN are constantly expanded and improved, new services are now routinely developed in the context of the AIN call model. An example of such a new service is taught in U.S. Patent No. 5,377,186, which was issued on December 27, 1994 to Wegner et al. It provides a system and method to recover the increased subscriber services from a database and deliver those services to subscribers of the PSTN, without requiring the improvement of the local switches to operate with the communications protocol of the Capabilities Implementation Part of Transaction / Advanced Intelligent Network (TCAP / AIN). A plurality of local switches that connect to an Intelligent Advanced Network (AIN) are enabled to provide subscribers with access to the network. This is accomplished by using at least one Virtual Services Switching Point (ViSSP) within the AIN to store a database of enhanced subscriber services. The local switches or tandem switches, to which they are linked, can recover the enhanced subscriber services from the ViSSP, which uses the Integrated Services Digital Network (ISUP) User Part of the SS7 call setup protocol. 'While the patent teaches the use of a virtual services switching point to expand AIN capabilities in local switches without costly improvements to these local switches, the use of the ViSSP is limited to the use according to the AIN call model, described above. Another example of one of such new services is taught in an article published by the Institute of Electrical and Electronic Engineers (XP 000269625) on June 23, 1991. The article is entitled "DELIVERY OF CALLS TO PORTABLE TELEPHONES REMOVED FROM HOME, USING THE LOCAL EXCHANGE NETWORK ", COMMUNICATIONS: ELEVATED AT HEIGHTS, and was presented by Beller MJ in Denver, on June 23-26, 1991. The article describes a method called" Crankbac method of call handling, with consultation of SCP ". According to the Crankback method, a call placed on a mobile phone is routed to the termination switch. This termination switch recognizes the number as being directed to a called party, which has an assigned and active call redirection feature. Upon receiving the IAM, the terminal switch issues a query to an SCP, which returns the call redirection number. The terminating switch then formulates a release message containing the new termination number and the originating switch uses the new termination number to complete the call as a standard call. The computer communications industry and telecommunications industries are beginning to melt, but they have always been a measure of difficulty with the integration of the two. The PSTN has been perceived by those in computer communications as a closed architecture, which encourages computer vendors to move the PSTN by creating overlapping networks that avoid capitalization on any of the core PSTN functionally. The call address, which uses an overlay network, requires many additional connections to the PSTN for both access and egress. In addition, the address within the PSTN of networks often overlapped to connections established in nodes that are redundant to the call path. In superimposed networks where there are limited connection points to the PSTN, calls can be directed over significant distances even through a call that can be completed at a network switching node where it originated. To solve this problem, the telephone industry has supported the release link trunk functionality solution for the subsequent address or redirection of the calls. The functionality of the release link trunk may reside in either the PSTN or in an overlay network, but the release link feature may only reside within a network switching node. This release link feature allows the data message, usually the SS7 ISUP message, to release a call back to the call set-up point, where the redirection can be made for this call. This feature is widely used in the current telecommunications industry. Although the release link feature resolves some of the problems associated with redundant connections in the call path, the fact of switching the resident node is a significant drawback. The development of switching and differences in proprietary protocols limit the nature and availability of release link characteristics. The development of new features for the switching nodes of telephones generally requires many months and involves considerable expense. Therefore, the switching vendors give priority to the performance of the characteristic based on the aggregate demand, the frequent placement of service providers in a waiting position for the realization of characteristics. Attached solutions, such as Intelligent Peripherals (IP) are, therefore, often sought after. Smart peripherals provide management of device resources, such as voice response units., voice announcers and DMTF units for call activated services. IPs are not seen in the signaling network, however, so they are not able to directly control call processes. Other novel related jobs have been inspired by the unique requirements of call centers, for example. In the activation of sessions, the transfer and connection, the line transfer and other features have been carried out in new solutions using the PSTN. However, until now, service providers have not been really effective in the ability to direct calls resident in the PSTN.

SUMMARY OF THE INVENTION It is an object of the invention to provide a method and apparatus for improving the ability to recall in the PST, by the separation of the call model from the AIN, while adhering to the basic principles and requirements of the common protocol. Channel signaling It is a further object of the invention to provide a method for dynamically directing selected calls through an intelligent switched telephone network, which uses the common channel signaling messages, generated by the call control node in the path of the common channel signaling control for a call. It is still another object of the invention to provide a method for completing calls through the PSTN, in which the calling party controls the subsequent address of the call by the interactive selection of the parameters of a source in a call connection. It is a further object of this invention to provide a method for completing calls through the PSTN, in which the called party controls the subsequent address of the call by redirecting the call to the other called party. It is a further object of the invention to transfer the call control from the PSTN to a virtual switching node, which is a virtual node in the switching plane of the PSTN and a physical node in the signaling plane of the network. It is still another object of the invention to provide control over the call direction, using a data network for communications, with a call control node in the signaling plane of the PSTN, to increase the services of the subscriber. The invention provides a method for dynamically guiding selected calls through an intelligent switched telephone network, which uses the common channel signaling messages, generated by a call control node in the common channel signaling control path for the call, which includes the steps of guiding a call through the network to a first termination, in response to a number dialed by a calling party, so that the common channel signaling path for the channel passes through the node ( 208), CHARACTERIZED because: the call control nodes (208, 232) include a channel signaling interface (302), to receive and send common channel signaling messages related to the calls selected therein guided by the routes and links establishments, which direct all common channel signaling messages, associated with the selected calls to the control node l of calls; elements for examining the common channel signaling messages and transparently forwarding the selected common channel signaling messages to an adjacent signaling node (34, 36); elements for generating the common channel signaling messages in response to predetermined criteria; elements for guiding individual calls, virtually switched over thereto and a data interface (306) for receiving data inputs from a network (226) not associated with the signaling network of the common channel; and the call control node assumes control of the selected calls, and determines a new termination for each selected call, the new termination is indicated directly or indirectly by the predefined criteria, and initiates a release of the first termination, using at least a common channel signaling message, without releasing the calling party (27); and initiates the network? rection_ of the call to the new termination, using at least one other common channel signaling message. The invention also provides an apparatus (208, 232) for dynamically guiding selected calls through an intelligent switched telephone network, comprising a common channel signaling interface (302), for receiving messages from and sending messages to a network of common channel signaling, a memory (312) for storing at least one of the common channel signaling messages, a memory for storing programs (314), a processor (304) for examining the common channel signaling messages, received in the common channel signaling interface and for generating common channel signaling messages, for controlling the connections, CHARACTERIZED because: this apparatus (208, 232) further includes an element for transparently forwarding one of the common channel signaling messages to an adjacent signaling node (34, 36); elements for generating common channel signaling message in response to predetermined criteria; elements for guiding individual calls, virtually switched through it; and, a connection to a network (226) of data not associated with the common channel signaling network, to receive data inputs to control selected calls; and the programs enable the processor to guide calls routed virtually through the device, and evaluate predefined criteria to determine an action regarding the control of call connections. The invention further provides a system for dynamically directing selected calls through an intelligent switched telephone network, whereby calls are guided through a first termination for the call, so a signaling control path for the call passes. through a call control node (208, 232), CHARACTERIZED because: the 'call' control node is a virtual node in the network switching plane and a physical node in a common channel signaling plane the network and includes elements for examining common channel signaling messages and transparently forwarding one of the selected common channel signaling messages to an adjacent signaling node (34, 36); elements for generating common channel signaling messages, in response to predetermined criteria; elements to guide individual calls computed virtually through them; and a connection to the network (226) of data not associated with the common channel signaling network, to receive data inputs to control the selected calls; and this call control node is enabled to release the call in a forward direction from the call control node and reconnects the call to a new termination specified by the data from an external source received by means of the connection to the call network. data, without releasing a calling party (25) associated with the call. The method and apparatus, according to the invention, provides the tools for the efficiency of the switching power resident in the PSTN. They also provide the tools to develop and deploy, quickly and inexpensively, new services in the public switched telephone network. The method according to the invention provides a new model for developing telephone services where the calls are guided by a guide and link sets directing the common channel signaling call control messages to the call control node. Therefore, the control of the call is assumed by the call control node, the cal functions as a virtual switching point in the call path. The call control node may use signaling messages from the ISUP to control any connection point in a call path. This allows to establish and release individual connection points in the call path, while maintaining a connection with the calling party. The call control node is preferably a high speed computer machine, having a common channel signaling interface, a data network interface and application programs that allow the call control node to examine, modify and generate messages of common channel signaling. This mode of a call control node is named as the Virtual Switching Point (VSP) and is assigned to a Point Code of the switching node in the common channel signaling network. Alternatively, the call control node may be an Intelligent Signal Transfer Point (ISTP), programmed to perform the required functions of the call control node. It is considered that at least initially, only selected calls will be guided through the VSP, in order to simplify the application programs and scale the processing power of the computing machine required in the node. The initial development of a VPSP, therefore, preferably leads to a minimum conditioning of the network in which the line or trunk connections of the "SUP" are assigned or installed, to guide the selected calls to the call control node. The switching translation tables are created to guide the selected calls to one or more groups of designated ISUP line connections, hereinafter referred to as "enhanced ISUP trunk" or "EISUP trunk". The group of trunks of the EISUP can be a group of trunk of posterior loop or a group of trunk inter-commutators. In any case, the VSP is a virtual node between the terminating ends of the trunk group and a physical node in the signaling plane, so all the call signaling messages of common * channel, - related to the selected calls, they are physically guided through the VSP. This conditioning network can be located at a single switching point or a wide network made, depending on the type of service offered and the extension to the lime is made available to the public. After achieving the initial conditioning of the network, the rapid development and development of new services becomes possible in the upgraded portion of the network. If the ISTP is used to carry out the functions of a call control node, the only network conditioning required is the programming of all the ISTPs required to control the selected calls, making certain that the selected calls transit the line connections. or ISUP logs associated with the programmed ISTP. The method and apparatus, according to the invention, provides call control flexibility within the PSTN, hitherto not considered. Intelligent controlled logical connection points in the call path through manipulation of the Integrated Services ISDN User Part (ISUP) of the common channel signaling message conventions can provide flexibility significant added. While the invention deviates from the restrictions of the AIN call model, it provides effectiveness to the resident guidance power in the PSTN by adhering to the basic principles and requirements of the ISUP common channel signaling, as defined in the ANSI standard. TI.113.3 and Standard CCITT Q763.

BRIEF DESCRIPTION OF THE DRAWINGS The invention will now be further explained, by way of example only and with reference to the following drawings, in which: Figure la is a schematic diagram showing the layers of signaling protocol SS7; Figure lb is a schematic diagram of an SSUP ISUP message; Figure 2 shows a progression of a call between two local exchanges in the PSTN, using common channel signaling and call control methods of the prior art; Figure 3 is a schematic diagram, showing an example of the call flow of the AIN of the prior art; Figure 4a is a schematic diagram showing the architecture of the call control node referred to as a Virtual Switching Point (VSP) and its connection to a Smart Peripheral (IP) through a data network; Figure 4b shows a message format that can be used for the exchange of messages between the VSP and the IP; Figures 5a-5d are schematic diagrams of the network configurations, which can be used to carry out the calling process, according to the invention, in the PSTN; Figures 6a-6e are schematic diagrams showing the initial call set-up sequences, according to the methods of the invention; Figures 7a-7c are schematic diagrams showing the intermediate call release sequences used in the practice of the methods of the invention; and Figure 8 is a schematic diagram showing a final call release sequence, used in the practice of the methods of the invention.

DETAILED DESCRIPTION OF THE PREFERRED MODALITIES The Public Switched Telephone Network (PSTN) has developed an integrated automatic intelligent network, capable of directing sophisticated autonomous calls. The guided calls through the PSTN follow established call procedures. These calling procedures have been developed together with the capabilities of the PSTN. The call procedures determine how calls are handled in the network and consequently determine the type of services that may be available in the network. Call procedures have been developed to provide reliable interconnection between network elements through a rigid set of rules. These calling procedures are mainly incorporated into the standards for common channel signaling. Signaling System 7 (SS7) is the currently accepted standard for common channel signaling. This SS7 is a developed international standard or CCITT, now known as the Union of Industrial Phones (ITU). The Figure shows a schematic diagram of the SS7 layers. They consist of: the layer of the Message Transfer Part (MTP) used to establish the connectivity of the signaling link between the switching elements; the Signal Connection Control Part (SCCP), used for the control data link connections through the switched telephone network; the User Part of the Integrated Services Digital Network (ISUP), used to transmit the route information between the switching elements; and the Capacity Application Part of Transaction (TCAP), used to formulate database queries and provide database responses for smart phone services.

The SCCP, the ISUP and the TCAP are each components of the application layer of user parts of the SS7 protocol. The ISUP is the most common exchange guide protocol, used in the telephone network at this time. With the MTP established and the common channel signaling links synchronized and aligned, five basic ISUP messages are required for the handling of most calls. These messages include: Initial Address Message (IAM), - Address Completion Message (ACM); Response Message (ANS); Release Message (REL); and Liberation Release Message (RLC). In addition, under rule TR-NWT-000246, a suspended message is sent when the called party disconnects first, but this is not universally used in the SS7 network. In the ISUP signaling, the Message Signal Units (MSU) are the SS7 messages used to carry the call guidance information between nodes. Figure lb is a schematic diagram of the ISUP message format. Each MSU 80 includes flags to indicate the start and end of the message, as well as the sequence numbers back and forth, to facilitate the ordering and grouping of the message units. Each MSU 80 also includes the Signaling Information Field (SIF) which consists of a guide tag 83 and an ISUP message 84. The guide tag 83 stores an Organization Point Code (OPC) and a Destination Point Code (DPC) for the message. Point codes identify only each node in the SS7 memory. Part 84 of the ISUP message is variable in length and includes such information as the Circuit Identification Code (CIC), which identifies the trunk of voice over which the call is carried. A Message Type 88 that indicates the type of message of the ISUP is sent and details about the call are contained in the Mandatory Fixed Part (MVP) 92 and an Optional Part (OP) 94. To illustrate the use of the message of The ISUP for call connection control, Figure 2 shows a progression of a call between two local exchanges, 34, 36, which use the common channel signaling network to establish and control the call. When the caller A picks up the telephone 25, the user's line 27 is captured and the caller receives a dial tone, which indicates that local exchange is available to receive the dialed digits. Caller A dials 234-5678, for example, as indicated in Figure IB. Upon receiving the dialed digits, the local exchange 34 consults its translation tables and determines that it can not complete the call locally. These translation tables indicate that the call should be guided to the local exchange 36. In response, the exchange 34 formulates an ISUP IAM message for a call from the regular PSTN. The local exchange 34 inserts its point code (000) into the Source Point Code (OPC) of the route tag 83 (see Figure lb) and the point code of the local exchange 36 (002), in the Code Destination Point (DPC) of the guide tag 83 and forward the message on the A50 links of the SS7 to a Signal Transfer Point (STP) in a corresponding pair of the STPE 72. 'The STPs examine the DCP to determine that the message should be sent forward to a local exchange 36. Upon receiving the ISUP IAM message, the local exchange extracts the dialed number and consults its translation tables, which indicate that the call will be completed to the party B call on the telephone 41. Therefore, it captures a member of the ISUP trunk group 38 indicated by the Circuit Identification Code (CIC) 86 of the ISUP IAM message and places the ringing signals on the telephone line 41 of the called party. The local exchange 36 then formulates an ACUP ISUP message indicating that the address is complete and returns the ACM message via STP of the STP pair 72 to the local exchange 34, which generates a ring signal on the user's line 27 for the benefit of the caller A on the telephone 25. When the called party B answers the telephone 41, the off-hook condition is detected by the local exchange 36, which formulates an ISUP ANS message that is sent forward by means of an STP of the STP pair 72 on the A 50 links of SS7 to the local exchange 34. Upon receiving A's message, the local exchange 34 ceases the ringing signal on the telephone 25 and continues the conversation between the caller A and the party call B on the member of group 38 of the trunk of ISUP. After finishing the conversation between the parties, caller A hangs telephone 25, which sends a hang indication to local exchange 34, which indicates that the call has been disconnected. Local exchange 34, therefore, formulates an ISUP REL message that is sent forward via STP of STP pair 72 to local exchange 36. Upon receiving the REL message, local exchange 36 applies the tone of marked to the telephone line 41 of the caller, indicating that the call has been disconnected. When the called party B hangs up the telephone receiver 41, the hung condition alerts the local exchange D that the call was disconnected and releases the captured number from trunk 38 of the ISUP. Next, it formulates an ISUP RLC message which returns to the local exchange 34 and this local exchange 34 releases the originating end of the ISUP trunk member used by the call. The RLC message is not activated by the disconnection of the called party B. Figure 3 is a schematic diagram of a call process based on the prior art AIN call model, which is performed extensively in the PSTN of North America In the standardized activators of the AIN embedded in the factory of the Service Switching Points (SSP), they make it possible for the SSPs to ask the questions to the TCAP to the databases, when an activation condition is satisfied. The AIN triggers can be based on 'dialed digits, the identification of the origin line or any of the various other variables recognized during the various points in the call connection process. In the example shown in Figure 3, when a caller 100 picks up and dials the number 1-800-777-7777, the SP 102 collects the dialed digits and consults their guidance tables, instructing them to forward these in a message of IAM through the network SS7 to SSP 104. The 1-800 marked in the IAM message activates the SSP 104 to formulate a question of the TCP that is sent to the AIN SCP 108, which translates the digits marked 1- 800-777-7777 at the physical address of the called party, which is a number from PSTN 613-751-0823. The AIN SCP returns the PSTN number in a TCAP response message to the SSP 104, which queries its translation tables to determine the area key "613" is served by some other switch in the switched network. Therefore, it formulates a message of the ISUP IAM that includes the number of the transferred PSTN and transmits that message in the switched network, which sends the message to the switched node, which serves the called number. The call is then terminated to the called number and the call progression with the associated ISUP messages follows, as described above with reference to Figure 2. The present inventors provide efficiency to the PSTN resident switching faculty by capitalizing the capabilities currently available in the standard ISUP signaling methods. This is achieved by treating the logical connection points in the call path, such as controllable connections, which can be manipulated within the ISUP message builders, to utilize the versatility embedded in the existing network for improved call handling. The invention thus provides a resource for rapidly developing and deploying call services by provision of a new network element, which operates in the network as a virtual switching point, pro not loaded by the switching factory or switching functionality. . The invention can be carried out in the network using any one or more of a number of options, which are explained below in detail.

In order to illustrate a service, which can be performed using the invention, a practical application involving an Intelligent Peripheral in the form of an Interactive Voice Response (IVR) unit will also be explained in detail. It should be understood that the description that follows is exemplary and that a vast array of services can be developed and deployed rapidly, using the methods and apparatus of the invention. In general terms, the invention relates to a method for performing flame redirection, during a telecommunications session, by collecting information from a calling party or a called party and using that information to dynamically redirect this call. Redirection is achieved using the signaling protocols and procedures of the ISUP signaling. Call control is achieved with a call control node, which is a virtual node in the network switching plane and a physical node in the signaling plane of the network. The control of the selected calls is passed to the call control node guiding selected calls to the telecommunications service facilities, designed to guide these selected calls to the call control node. Calls that do not require special control are guided over regular telephone installations.

This method has the advantage of allowing the control of a call from the call control node in the signaling path SS7 by the release of the call in any direction of entry or egress independently. The originating end of a controlled voice trunk of the SS7 is thus functionally separated from the terminating end of the trunk. With the signaling of the SSUP ISUP, the switching system at the originating end or the termination end of the trunk does not realize that the connection is not retained at the opposite end of the installation. This allows any end of the call to end at any other destination, without the loss of the connection. The invention can thus be used to exert more control over the calling route by allowing either the calling party or the called party to increase the flow of this call.

VSP System Architecture Figure 4a is a schematic diagram of an exemplary architecture for a Virtual Switching Point (VSP) 208. This VSP 208 is a preferred selection for the call control node, according to the invention. The VSO 208 includes a common channel signaling interface 302, the cal is capable of receiving common channel signaling messages from a common channel signaling network and broadcasting messages on the network. The common channel signaling interface 302 is accessed by a processor 304, which also has access to the data re-interface 306 which may be, for example, a TCP / IP interface for connection to a data network 226. , such as the Internet, or any other package transport. The processor 304 is connected to a data collector 310 to allow access to the RAM 312 and the ROM 314, as well as the disk storage memory 3126. Each RAM, ROM and disk storage are configured and used in the manner well understood in the art. Processor 304 also has access through data collector 310 to block 308 of ISUP. which allows the application programs executed by the processor 304 to examine, modify and generate the messages of the RAM ISUP. The ISUP block 308 may also be part of the RAM 312. Also connected to the data network 226 is an Intelligent Peripheral (IP) 214, which will be discussed in more detail below, as an example of a practical application of the invention in the PSTN. The IP 214 is connected to the PSTN via a Switching Point (SP) 36 and the data network 226, as will also be described below. Communications between the VSP 208 and the IP 214 through the data network 226 can be achieved according to any acceptable protocol. Figure 4b shows a packet 300 of data in the TCP / IP format and an exemplary data content of the packet, which will be referred to in the discussion that follows. Communications through the data network allows any programmed processor to communicate with the VSP 208 to direct the call flow by providing a new termination number, for example. Those skilled in the art will appreciate the potential and flexibility in the design of the service and the realization that this enables.

Network Configurations Figures 5a-5d schematically illustrate four preferred network configurations for carrying out the invention in a portion of the PSTN. These four network configurations are only exemplary. Other configurations can be used for the same purpose. In each of the network configurations described below, the network is configured to direct selected calls through a call control node, which may be the VSP 208 (see Figures 5a, 5c or 5d) or a Point of Intelligent Signal Transfer (ISTP) 232 (see Figure 2b) as explained below. The VSP 208 is the preferred call control node, due to its low cost, simple architecture and dedicated function. The ISTP 232 can be used, however, to carry out the invention as other intelligent processors can connect to the common channel signaling network of the PSTN and assign a point code as if it were a switching node or transfer point of signal in the network. In each of the network configurations, shown in Figures 5a-5d, a subscriber telephone 25 is connected to a network switching point 34 (hereinafter referred to as SP34) on the subscriber line 27. The SP 34 is connected to the common channel signaling network and equipped in at least with the signaling capability of the ISUP. The SP 34 is capable of generating, sending, receiving and examining the messages of the ISUP SS7. For the purposes of this document, the SP designates a network switching point as well as a network Service Switching Point (SSP), which is equipped to perform a TCAP query to an SCP. The SP 34 has a line side to which the subscriber line 27 is connected, and a trunk side from which a plurality of ISUP trunk, such as the ISUP trunk group 200, connects the SP 34 to other SPs In the net. For purposes of clarity, only one switching point in the PSTN is illustrated, an SP 36. Each SP 34, 36 is connected to the PSTN 202 in a manner well known in the art. The PSTN, in turn, includes a plurality of SPs that service a plurality of other subscribers.

In order to enable the invention to be practiced in the network, SPs selected in the network are either equipped with the EISUP trunks, which are designed to guide selected calls or with the posterior loop trunks 234 of a ISUP (see Figure 5C) that are reserved for that purpose. The trunks of the EISUP 212 and the rear loop trunks 234 of the ISUP are preferably groups of trunks of the ISUP carried over the facilities DS1 or El, which respectively accommodate 24 and 30 voice channels. Each voice channel is mentioned as a trunk member. The VSP 208 connects to the common channel signaling network in the STP pair 204 using links A 210 of SS7, for example. As will be explained below in some detail, the route translations in the SP 34 guide selected calls on the trunks 212 of EISUP 0 loops of posterior loop 234, which invoke messages from a 1SUP to be sent using the route sets and link sets associated with the trunk groups 212, 234 to the VSP 208, so that the SPs 34, 36, the VSP 208 appear to be a switching node in the path of call associated with those trunks. Since the VSP 208 serves as a virtual switching node in the call path, it is enabled to take control of a call by treating other connections in the call path, such as controllable connections that can be released or reconnected, as required, using the signaling messages of the ISUP, which it generates, manipulates or modifies as required. The SS7 guidance lines followed by the connection of the VSP -208 in the network conform to the ANSI Standard IT .111.5-1994, paragraph 7.2. It should be noted that for simplicity of illustration, the "Point Codes (PC) in the drawings attached to this document, do not follow the convention of assigning the point code, recommended in the ANSI Standard TI.118.8, where the code 254 network point has been assigned to AT & T, 253 to U.S. Sprint and 245 to Telecom Canada. In actual practice, the point codes are comprised of a network, cluster and member number, with the cluster numbers and members assigned by the specific bearer, where the network code has been assigned. Generally, members 0 and 1 are reserved for STPs. For ease of illustration, they have been used for SPs in the discussion that follows. A Smart Peripheral (IP 214) is incorporated into the network shown in Figures 5a-5d to illustrate the versatility of the route that can be achieved using the methods and apparatus according to the invention. As an example, the IP 214 can be an Interactive Voice Response (IVR) platform. The IP 214 is connected to the SP 36 using a trunk group DS 0, 216, which has coordinated control with the VSP 208. The coordination of the call route between the VSP 208 and the IP 214 is achieved using a 226 installation. of data network. Such a data installation can operate under any of a number of data protocols well known in the art, such as SR 3511 AIN IP. The data network 226 may be an X.25 network, the Internet, an ATM network, or the like. All calls handled by the VSP 208 are guided over trunk group 212 of the EISUP in a manner that will be explained below in greater detail. Figure 5b shows another network configuration that can be used to carry out the invention. In this configuration, the VSP 208 is replaced with similar functionality embedded in an Intelligent STP (ISTP) 232, as described in U.S. Patent No. 5,586,177, which was issued to Ferris et al. December 1996. Ferris et al. describe an apparatus for introducing aggregate functionality into the signaling network with an ISTP. The ISTP can be programmed to function as a call control node, according to the invention. With this network configuration, enhanced trunk groups of the ISUP are not required, because the call control messages are inevitably guided through one of the ISTP pair 232. Of course, it is necessary to program each ISTP in the pair identically. A disadvantage of using the ISTP 232 as a call control node is its relatively complex functionality as a Signal Transfer Point and the fact that it must efficiently handle a considerable message traffic load in addition to its functions as the control node of call. While this configuration potentially reduces the number of trunk groups required in the network, triggers and call control algorithms must be incorporated into the software (program) of the complex ISTP, which can introduce delays and costs similar to those associated with the software development for network switching points. The embodiments using the VSP 208, shown in Figures 2a, 2c and 2d, are therefore preferred. The calling route using the ISTP 232 will be explained in more detail below. Figure 5c shows * another network configuration suitable for carrying out the invention in a portion of the PSTN. All of the network elements shown in Figure 5c are the same as those shown in Figure 5a, with the exception that the EISUP trunks 212 are replaced by the rear loop ISUP trunks 234. These rear loop trunks 234 are connected to the SP 36. This configuration consolidates the selected calls controlled by the call control node, according to the invention, on the ISUP rear loop trunks 234, providing a centralized configuration for guiding calls. selected. The VSP 208 operates in substantially the same way as with any other network configuration. Selected calls are routed through the translation tables to the back loop trunks 238, connected to the SP 36, which causes the SP 36 to generate an ISUP IAM message, which is sent forward to the VSP 208, as will be explained below in more detail. Figure 6d shows yet another network configuration for carrying out the invention. All network elements shown in Figure 5d are identical to those shown in Figure 5a, with the exception that IP 214 is not connected directly to SP 36. Instead, IP 214 is connected to another switch similar in PSTN 202. This architecture is illustrated because it introduces certain complications into the message associations that are easily shared with them, but require special consideration. Practical solutions are indicated in the discussion that follows. In all other aspects, the network configuration shown in Figure 5d is the same as the network configuration shown in Figure 5a.

Message Transfer Part Network Conditioner Two basic b configurations for implementing the MTP, of the SS7 can be used to carry out the invention in the PSTN: 1) a call control node, which is assigned to a point code and it appears on the network as a switching node to cause any call control message to guide that node from the SS7 network, as disclosed in U.S. Patent No. 5,377,186. If the posterior loop trunks 234 are used, two point codes can be assigned to the VSP 208 to allow the same CIC to be assigned to each end of each subsequent loop trunk member, as will be understood by those skilled in the art. Alternatively, if the SP 36 (see Figure 5c) supports multiple point codes, two point codes can be assigned, to the SP 36, allowing the ends of the rear loop trunk member to end at different point codes in the SP 36. As will also be understood by experts in the field. As a further alternative, different CIC codes may be assigned to each end of each subsequent loop trunk member, such that the VSP 208 may be programmed to compute an CIC when it responds to an ISUP message. For example, the opposite ends of a back loop trunk member can be assigned to CIC in sequence and the VSP 208 can be programmed to increase the CIC in the ISUP message by one, as will also be understood by experts in the matter. Other algorithms can also be used for CIC assignments for posterior loop trunk members. The communication of the SS7 with the VSP 208 can also be achieved using F links, in this case, the rear loop trunks 234 will be the most practical embodiment for the trunk groups of the ISUP designated in the SP: The guidance methods described above for the back loop trunks they can be used to control the trunks of the posterior loop ISUP. However, this option is not preferred because the implementation to guide the ISUP is identical to the first option and is efficiently sacrificed. In addition, if the F links are used, an F link connection to a call control node is required from each SP / SSP in the lime the invention is to be carried out.

Establishment of the ISUP Trunk Figures 5a and 5d illustrate the configurations in which the trunk groups of the inter-switch EISUP are established. Physical trunk links are established between SP 34 and SP 36. Virtual trunk links are established between SP 34 and VSP 208 by creating a set of routes that direct all selected outbound calls to the trunk group of the trunk. Increased ISUP, virtually connected to VSP 208. A corresponding virtual trunk group is established between VSP 208 and SP 36, creating a set of similar routes between VSP 208 and SP 36. The virtual trunk groups are set to much the same way for the posterior loop trunks 234 (see Figure 5c), where the virtual trunk groups are established between each end of a group of posterior loop trunks in the SP 36 and the VSP 208. YES, as as described above, two point codes are used in either SP 36 or VSP 208, as will be explained below in more detail, two sets of routes to VSP 208 are required. Alternatively, CIC codes can be increased using an appropriate algorithm, as defined above and explained below in more detail. In that case, a single set of routes can be used.

When the VSP 208 is configured in the network as a virtual node using any of the increased ISUP trunk group configuration or the rear loop trunk group configuration, the trunk groups must be established as a part of the conditioning of network. The logic, therefore, is required in the VSP 208 to allow certain network signaling messages to pass transparently through the VSP 208.

Example 1 When a continuity message is sent during the initial upward turn of a trunk or during the capture of a trunk in the establishment of a call, an action of sending back tones or a loop to an installation is required at the far end of the physical termination. This requires the VSP 208 to modify the point codes in the message and pass the message without further modifications to invoke the expected action - at the terminating end of the physical trunk. Failure to perform this action will result in the generation of a trunk condition that has failed at the trunk's origin end, which would cause the trunk to be removed from service. The VSP 208 must modify the point codes (and, in some cases, the CIC code, as explained above) so that the forward message will arrive at the switch at the trunk termination end, to allow the action required for the tumbling ascending of the captured trunk.

Example 2 Blocking and unblocking messages must be transparently passed through the VSP 208, again with the point code and, in some cases, CIC modifications, as required, to ensure conditions in which both ends of the facilities physical are aligned. Only certain messages required for increased call control will be changed substantially by the VSP 208.

Path Transfers In order to practice the invention the address of selected calls must be controlled by the call control node. If this call control node is a VSP 208, another component of the network conditioning, required to enable the use of the invention, is certain changes to the address translation tables in the PSTN SPs where the invention is will realize. The standard translation methods and the standard translation tables can be used to perform the address requirements of the invention.

When a telephone number is dialed in an AIN network, the dialed digits are examined to determine how to guide the call. The route decision is made by consulting the address tables, which direct the switching system to select a trunk group specific output for the call, if this call is not a part called with service by the switching system. Under certain conditions, a dialed number will activate an AIN switching system to formulate a TCAP message containing the dialed digits and will dispatch the TCAP message to a database for a conversion of the dialed digits to another number. Upon receiving the converted number, the switching system consults its address tables to determine a trunk selection for the call. When the network is conditioned to practice the invention, therefore, it is necessary to create entries in the address tables of the enabled SPs, to ensure that the selected calls are routed to a group of logs, which has a virtual connection to the VSP 208 The route set of the SS7 associated with the trunk group will direct signaling messages related to the call through the VSP 208, which will allow the VSP 208 to control the call immediately, as will be explained below.

The selected calls will normally be guided to the call control node, based on the dialed digits or a translation of the dialed digits. They can also be guided to the call control mode using the prior art translation methods, based on the identification of the call-line to allow special services to be developed for individual subscribers, companies, corporations or the like.

Trunk Selection Criteria - Reflection Trunks that involve reflection, which can be captured from any direction and is a condition that occurs when each end of a specific trunk is captured at the same time. The reflection is a potential problem with the groups of trunks of the EISUP inter-switches, if the log group is equipped for outgoing calls in each SP. This problem is easily solved by designating at least two groups of interchangeable EISUP trunks and equipping the trunk groups as output trunks only from the respective switches. This is the simplest option for the elimination of the reflection. However, it is advantageous from the perspective of network utilization, to use two-way trunks, which can be captured from any SP. There are several known switching path algorithms, which reduce the anticipated reflex by causing different trunks to be selected at each end of a group of trunks, clockwise, or counter-clockwise being a good example. Using this algorithm, the last inactive trunk is located, which must be the same trunk member in each SP, and one stage of the SP clockwise goes to the next trunk available, while other stages of SP go against the direction of the clock to the next available trunk, when an inactive trunk is selected for a new call. If both ends of the same trunk member are captured at the same time, a convention is used to determine which switching system is required to be stepped down and select gold trunk member. The VSP 208 can be • in the control office and both SP 34 and SP 36 are forced to descend, but this situation is handled more efficiently as other management emergencies where messages are transparently passed through the VSP 208 and any SP 34 o SP 36 is dominant and the subordinate office goes down.

Number Portability Number portability is being or has been performed on most PSTNs. This number portability may result in a situation in which the dialed number is a ported number or the VSP 203 attempts to redirect a call to a ported number. In certain embodiments of number portability, a terminating switch will return to a Release With Cause = "LNP" or the like, which attempts to warn that the originating switch that the number is a ported number, and that a query is required to the database to retrieve the physical address of the called art. In this case, the Release with Cause can not be passed back to the originating switch, because its records show that it has a rising call completed and will not have procedures to share with the reception of the message. The VSP 208, therefore, must be enabled to directly share with number portability. Accordingly, the VSP 208 may be equipped with any TCAP capability, or be enabled for the query SCPs using its data network interface 306 (see Figure 4a) and the protocol, such as the SR 3511 AIN IP, or similar, as will be understood by the experts in the field. Alternatively, translations are available, which indicate adjacent switching capabilities. This provides a mechanism to allow TCAP enabled SPs to perform number portability queries at a point before the VSP in the release sequence.

Billing Requirements Billing is also a consideration for enhanced services and many special service calls are billed based on call volumes. The overall duration of a total call will be available at the originating SP (34, 36) and the existing free long distance call billing routines, LAMA, can be used when appropriate. If subsequent call information is required, that information may only be available to VSP 208 and SP (34, 36) at the terminating end of trunks 212 of EISUP, when groups of posterior azo logs are not used to guide selected calls. In these circumstances, there are at least two viable options for billing. 1) Billing in the SP: The existing billing options, available in the SP at the terminating end of the ISUP, may be able to generate billing records with appropriate modules attached to identify types of calls with specific information. If many modules do not exist, switch modifications may be required to allow billing to be performed in that SP. The advantage of billing in the SP is that most SPs already interface with the billing platforms and support the expected formats. 2) VSP Billing: The VSP 208 handles selected guided calls through the network over the EISUP trunks and has access to all information related to duration, called party, calling party (when available), redirected number, etc. . Therefore the VSP 208 has all the information required to generate the billing records and can be programmed to compile the call records in any Bellcore AMA Format record (BAF), which may be required. Billing algorithms are well known in the art and there are many examples for IP or SP billing. The billing number must be associated with the subscriber and can be inserted in the field of the charge number of an IAM.

Form of Realization Control and Alignment of Messages In real-world embodiments of the invention, the VSP 208 may be required to handle tens or hundreds of selected calls concurrently. One purpose of this invention is to allow a new level of call flexibility by enabling a call to be redirected from one calling party to another, any number of times without disconnection of the calling party. In order to achieve this, the call control node must have at its disposal, at all times, a unique identification of each call in progress. Also, the call redirection instructions will normally be received from an external source, such as a data message received through the data network interface 306 (see Figure 4a). It is therefore important to coordinate the release and restoration of the egress circuits under the control of the VSP, 208. This is required to ensure that the correct exit trunk is released in the SP at the terminus end of the EISUP trunk to allow that a call is redirected to a subsequent destination. In order to control concurrent calls, the VSP 208 must be provided with a mechanism for the call guidance, which allows determining with which call any signaling message of the given ISUP is associated, and which trunk member is the SP at the terminating end of the egress trunk group is selected for the call. This requirement is complicated by the fact that Automatic Number Identification (ANI), normally found in the calling party's field of ISUP messages, is not always available. For example, some PEXs do not supply the ANI, or provide the same ANI for all calls. There are several options for coordinating and guiding messages and these options can be adapted depending on the configuration of the installation to which the service is provided.

Simple Terminations Associated with Individual Directory Numbers This option is the easiest to control and perform and can be used in configurations where each termination within a switched telephone network is probably the only one. This option can be used for certain service applications, however, because in the network two or more callers can be connected to the same number (one PBX, for example), at the same time. However, there are services for which this option can be used. For example, applications for improved call handling, where each agent has a different phone number that is associated exclusively with that agent, as in the case of certain call center, the coordination of data messages with calls in progress it is achieved by the correspondence of the data addresses (TCP / IP addresses) and the address of the telephone network (called party number).

TABLE I There is a direct correlation between the number of the Directory associated with each member of the called party, the Calling Party Number of the ISUP, GR 246 TI.113.3 and the TCP / IP address to or from which associated data messages are delivered. or received. Or Centrex or individual telephone lines can be used with this call guidance method. This method can not be used to guide calls associated with an IP because it is assumed that more than one line is required to the IP service and thus multiple calls would end in the same number. The TCP / IP address shown in Table I can be a dedicated address on the Internet or an intranet, or where a TCP / IP address is shared, a unique user or station identification can be attached to the message. The data packets containing the caller information can be delivered to the same location as the voice component of the call, once the TCP / IP address associated with the call is returned to the VSP 208 with sufficient information to allow the VSP 208 matches the TCP / IP address with a call in progress. In this example, the called number would be sufficient. There are many companies with products that coordinate the delivery of voice calls and data information related to the same location. This invention capitalizes and increases your work.

Individual Marked Numbers Associated with Termination Groups If the apparatus and methods of the invention are used to augment the services provided to a call center operated from a PBX or an Automatic Call Distributor (ADC), for example, or a Smart Peripheral, such as a V, where a dialed number can end a group of logs, the call guide can be achieved by assigning a block of numbers to the termination, which is equal to the number of facilities to handle the calls. Upon receiving a call in which the dialed number indicates such termination, the VSP 208 selects the next available number from the allocated block of numbers and inserts it into the field of the dialed number in place of the original dialed number, forward of the IAM. If all available numbers are in service, the VSP 208 immediately returns a message from the REL to the originating SP and drops or places the call in a waiting queue. Otherwise, the VSP writes a call control table entry and sends forward the IAM containing the new dialed number. Table II shows an example of a call control table, which can be used in this situation.

TABLE II The Intelligent peripheral or Call Center is provided with the Dial Number Identification Service (DNIS) and the terminating SP extracts the dialed number from the IAM and sends it forward on the termination trunk. The dialed number can then be returned to the VSP 208 in a TCP / IP message that allows the VSP 208 to coordinate the TCP / IP address with the dialed number, so the VSP will know which call will be redirected if it receives a message Redirect from that TCP / IP address. If the TCP / IP address is shared, the dialed number must be saved and returned along with the redirection instructions to allow the VSP 208 to redirect the call.

Number of Increased Calls for CLIDs Duplicates for Group Terminations This is an alternative solution to the problem discussed immediately above, where a single telephone number ends at a plurality of terminations on one or more trunks and the called number can not be used to uniquely identify a call. With this option, the Caller Line Identification is used to uniquely identify each call. This option capitalizes a unique CLID where calls are from the unique locations of the calling party, identified by a unique address of the calling party in a switched network. The CLID is used to correspond to a physical call with data information about that call. Certain calls do not provide unique CLID information, however. Calls that do not provide unique location identifiers (CLID) include PBX locations, where extensions are used, office calls not equipped with SS7, key systems or private numbers (private numbers can not be passed outside the network PSTN). With calls from these locations there is the potential to receive what appears to the VSP 208 as calls begged. The provision of a unique reference number for these calls is achieved by increasing the CLID by a predetermined value, such as 1, for each occurrence of the duplicated CLID. The increased number can be reset after it reaches a predetermined value, which is large enough to ensure that duplications do not occur. Table III shows an example of a call control table that can be used to carry out this solution.

TABLE III See the previous discussion for the TCP / IP address treatment.

Examples of Selected Calling Guide In order to illustrate the calling route using the apparatus and the methods according to the invention, several examples of address are provided in the following description. An IP in the form of an IVR is used for the purpose of illustrating a simple service that can be developed. As noted above, it will be readily appreciated by persons skilled in the art that the invention can be used to carry out a vast array of services. For purposes of clarity in the drawings, call set-up, mid-call processing, and call disconnection sequences are shown in separate drawings, and to limit duplicate replication sequences are shown in only one drawing, as will be explained below. . Figure 6a shows Caller A on telephone 25, annex to either SP 34 or switched network 202 on subscriber line 27. In response to an ad in a local newspaper, Caller A brand 555 GOLF (555-4653). The digits can be received by the SP 34 or from the appearance of line in the SP 34 or from the switched network 202. The 555-4653 can also be a transferred number, received by the SP 34, transferred, for example, from 1 -800-555-GOLF, which is a free long distance number to the caller, without a physical termination associated directly in the network. Examples of the process of calling the number 800 are described in U.S. Patent No. 4,191,860 to Weber. Standard translators of the switch are used to cross the switched telephone network 202 following the standard guide conventions used in the network and known to those skilled in the art. The North American Number plan is the most common number plan in use within the PSTN, but other address identification options, such as trunk and trunk identification are also used for call guidance. The guide translations in SP 34 are structured to recognize 555 as a selected call and the call is routed to trunk group EISUP 212, reserved for such selected calls. Standard route conventions are used to achieve this. Other guided calls to SP 36 are guided using regular 2000 ISUP logs. Group 212 of trunk EISUP can also be used as an overflow trunk group for regular calls. To complete the selected call, 555-4653, SP 34 searches for an inactive trunk member in trunk group 212 and captures CIC 18, which it finds will be inactive. SP 34 then assembles the Initial Address Message (IAM) of the ISUP with a Point of Origin Code (OPC) 0 0 0, a Destination Point Code (DPC) 0 0 1, and a CIC 18, which identifies the next inactive trunk available in a virtual trunk group between SP 34 and VSP 208. The IAM message of the ISUP is sent forward to the VSP 208 by the SS7 signaling links 206, between the SP 34 and the STP pair. 205 and in the SS7 signaling links 210, between the STP pair 204 and the VSP 208. That VSP 208 extracts the number 555-4653 from the IAM message and uses the number in a look-up table to determine the service to the which the caller has access to. The table indicates that the number 555-4653 is to be moved to 777-4653, the number associated with IP 214. Table IV shows a translation table of an exemplary marked number.

TABLE IV In the translation table of the dialed number, the called number received is located in the table. All or a portion of the calling line identification (CLID) can be used in the translation, for example, to guide regional or similar calls. The dialed number can, therefore, have one or more entries in the table. Other variables, such as the time of the day and / or the day of the week, can also be used in the translation. There are many other options to guide that may require other variable fields not illustrated in the table. When the variables all match in Table IV, a Service Number and a Service Code are located. The Service number determines the termination for the call and the Service Code is used in subsequent guidance decisions, as will be explained below. The VSP modifies the number called in the current IAM message to the service number and uses the service code to determine the guidance to be applied to the ISUP message. Table V shows an exemplary common channel signaling translation control table.

TABLE V The common channel signaling control table allows a call control mode to execute the message address and the guidance function. The service code is derived from the translation table of the dialed number, as described above. The type of message is extracted from the field of the message type, the part that enters identifies where the message originated and where the subsequent messages will be returned. The part that leaves indicates where the message will be sent and from where the subsequent messages will be received. The common channel signaling control table helps keep the service logic in the call control node as generic and flexible as possible. Other variables can be added to the table, if required. VSP 208 uses the common channel signaling control table to determine that the OPC of the message should be changed to 001 and the DPC should be changed to the DPC associated with SP 36, PC 0 0 2. An entry of the table of Call control for the call is then created using any of the options described above in the Call Control and Message Alignment section to uniquely identify the call. The call control table is maintained dynamically on a per-call basis and supplies the details in the current call status for each call in progress. A preferred structure for the call control table is shown below.

TABLE VI Service Number: Each call guided to a termination is assigned a serial number, which is used to coordinate the call control messages. CLID: Can be used to identify the initial call, as described above, and is required in subsequent calls for billing information.

Common channel signaling point codes: The stored OPCs are associated with the source portion of the EISUP trunk. The DPC is associated with the trunk termination portion of EISUP. The CIC identifies the number of the trunk group that is used for the call. The combination of this information is required for subsequent actions in the call, as controlled by the call control node. The service number is the number which is used to guide a destination. It is derived from the translation table of the dialed number when an IAM is received at the call control node. It is normally supplied from an external source, such as a data network, after an initial termination for a call to be established. The action code is the call control information received from the switched network or via the data network, to drive the next action taken by the call control mode. The TCI / IP address is used for any communication between the call control node and the call termination. It is updated each time the service number is changed, and it can be blank. The VSP 208 collects the IAM message and issues it on the common channel signaling network, where it is sent forward on the guidance set between the VSP 208 and SP 36. The SP 36 receives the IAM associated with the CIC 18 from VSP 208 and used the regular translation tables and procedures to guide 777-4653, which is well known in the art. The translation tables instruct SP 36 to reserve line 216, which connects to IP 214, in this example, an ISDN line. SP 36 sends forward the information associated with the call to IP 214 in an ISDN establishment message. The parameters of the IAM are mapped directly in the ISDN establishment message, allowing the information contained in the message to be used to define the service to be applied in IP 214. The ISDN call processed, alerts and connects messages that I read are sent from IP 214 to SP 36m that translates the messages in a Complete Address Message (ACM) of the ISUP and Response Message (ANM) that are sent back to the VSP 208. While the speech path is being established, a call alignment message is generated by the IP 214. This alignment message preferably includes the dialed number, CLID and the IP address. The data message is transmitted on the data line 218 to the data network 226 from the IP 214 and sent forward from the data network 226 on the data line 228 to the VSP 208. Upon receiving the alignment message, the VSP 208 places the corresponding entry in the call control table, described above, and upon finding a match generates a unique serial number that is written to the table and returned to IP 214 in a recognition message. The serial number is then associated with the balance of the call, regardless of the number of terminations in sequence to other numbers. as will be described below. This exchange of information establishes the data link for the subsequent release and transfer of the call, the subsequent guidance is required. The messages may be provided in format, as shown in Figure 4b, where the data portion 3021 of the TCP / IP message 300 includes the Serial Number, CLID, Service Number, TCP / IP address, and a number. reconnected at times when call redirection is required. Upon receiving the VSP 208 of the ACM and ANM messages from the SP 36, a record is preferably made that the call has been answered and the ACM and ANM messages are modified, as required, with the use of the control table Common channel signaling and sent forward to SP 34, which provides the full path and allows the call to begin an interactive session with IP 214.

Network Switched Through VSP to IP Figure 6b shows yet another configuration for a switched network to complete the selected telephone calls, where the guidance through the PSTN is required to reach IP 214. As described above in the example prior, the SP 34 assembles an ISUP IAM with the Point of Origin Code (OPC) 0 0 0, a Destination Point Code (DPC) 0 0 1, and a CIC 18, which identifies the first trunk currently available in the group of trunks 212 of virtual EISUP, between SP 34 and VSP 208. The called number sent forward to the IAM of the ISUP is, for example, 555-4653. The IAM message of the "ISUP is sent forward to the VSP 208 via the signaling links of SS /, 206, between the SP 34 and the pair of STP 204 and on the signaling links of the SS7 210, between a couple of STP 204 and VSP 208. From the IAM message, VSP 208 determines using the called number in a look-up table of the translation of the dialed number, the subscriber and the service which the caller is accessing. That table also converts the number called 555-4653 to the service number 777-4653, the destination number associated with the IP 214. and the VSP 208 changes the called number in the current IAM message accordingly. The common channel signaling table indicates that the OPC must be changed to the PC of the VSP 208 (PC 0 0 1) and the DPC must be changed to the PC of the SP 36, (PC 0 0 2). An entry of the call control table is created for the call, as described above. The VSP 208 then collects the IAM message and forwards the recently collected IAM on signaling links 206 to SP 36. This SP 36 receives the IAM associated with the CIC 28 and applies regular translations to guide the so-called converted number, 777- 4653 The translation tables instruct SP 36 to guide the call through PSTN 202 to an SP that serves IP 214. SP 36, therefore, reserves an output trunk connected to PSTN 202, as specified in translation tables, and modifies the IAM to include its point code in the OPC and the point code specified by the guide set for the trunk group reserved in the DPC. The CIC of the captured trunk is inserted into the CIC of the AMI. This IAM is then issued on the SS7 network, where it reaches, directly or indirectly, the SP that serves the IP 214. In the SP that serves the IP 214, the parameters of the AMI are mapped directly in an established message of the ISDN, as described above. The information contained in the message is used by IP 214 to define the service to be supplied to the calling party. The ISDN call proceeds, alerting and connecting messages that then return from IP 214 to the SP, which it serves. This SP translates the messages into the complete address messages (ACM) of the ISUP and of the response (ANM) which, in turn, send through the switched network to the SP 36 and to the VSP 208. While the speech path is being established through a PSTN,. an alignment message is sent from IP 214 to VSP 208 via data network 226. This exchange of information supplies or confirms the IP address of IP 214 to VSP 208, to enable subsequent release and transfer of the call if the subsequent guidance is required, and supplies IOP 214 with a unique serial number identifier for the. call, as described above. The format of the message preferably conforms to one of the options described in the Call Control and Message Alignment section. Upon receipt of messages from ISUPO, ACM and ANM, in VSP 208, a registration entry is written that the call has been answered and messages from ACM and ANM were modified as described above and sent forward to SP 34, that completes a voice path, which allows the interaction between the calling party and the IP 214 that begins.

VSP to IP Address, Using Logs Around the Loop Figure 6c is a simplified diagram showing the call setup process for the network couration in which the VSP 208 serves as the call control node for an equipped SP 36 with trunks 234 of posterior loop of the ISUP. As explained above in more detail, the invention can be performed using a network couration, in which one or more of the SPs in the network are equipped with the ISUP backbone trunks groups to guide selected calls. The function of the posterior loop trunk groups of the ISUP is to guide the selected calls through the call control node, in this case the VSP 208. This is achieved using the standard translation tables and the methods in a well way. known in art. The calling process is performed very similarly to the examples described above, where when caller A dials a number, such as 555-4653, whose digits can be received from a line appearance 27 in SP 36 or can be received at an IAM message from the ISUP received in SP 36 through the PSTN 202. In any case, the translation tables in the SP 36 guide the call to an exit end of the ISUP rear loop trunks 234 and the SP 36 reserves an inactive trunk member of the group. A set of routes associated with the trunk group directs SP 36 to formulate an IAM having an OPC of 000 and a DPC of 001. SP 36 compiles the IAM of the ISUP and outputs it in the SS network / over the links At 206, where it is transmitted to the STP pair 204 and forward on the A 210 links to the VSP 208.

This VSP 208 handles the message as described above, using the lookup tables to determine the service number and service code associated with the service which the caller has access to. The common channel signaling control table also instructs the VSP to change the OPC in the IAM to the DPC and the DPC to the OPC. That look-up table also provides an appropriate CIC, so that the entry end of the posterior loop trunk member reserved by SP 36 is captured at the reception of the IAM. The dialed digits can also be converted during this process, depending on the access service, as described above. VSP 208 compiles the modified AMI and sends it back to the SS7 network where it is returned via links A 210 and 206 to SP 36. This SP 36 receives the IAM as an incoming call request and captures the end of entry of the posterior link trunks of the ISUP indicated by the IAM CIC. SP 36 then consults its guidance tables to determine when the call should be guided. In this case, SP 36 determines that the call must be routed to IP 214, which serves. This IP 214 is connected by an ISDN link 216 to the SP 36 and the ISUP call parameters are mapped to an ISDN facility, as described above. Also described above, the ISDN call proceeds and alerts and connects messages that are generated by the IP 214. An alignment message is also exchanged between the IP 214 and the VSP 228 by the data network 226. Concurrently, the SP 36 maps parameters from the ISDN, which the call proceeds, alerting and connecting messages in the ISUP, ACM and ANM messages, which are dispatched to the VSP 208 via the STP pair 204. The VSP 208 again consults the verification tables and reverses the point codes as described above and adjusts the CIC accordingly. The ACM and ANM messages are then dispatched back to the SP 36 to complete the voice circuit between the calling party and the IP 214, so that the interaction between the two can proceed, as will be described later in greater detail. Similarly, if the IP 214 does not connect directly to the SP 36, but connects to another switch in the PSTN 202, the calling process will be performed as described above, with the exception that rather than addressing the IP 214 directly, the SP 36 will consult its guidance tables and compile the IAM message accordingly, before issuing it in the PSTN, where it would be sent forward to the switch serving IP 14, as described above with reference to Figure 6b.

ISTP as a Call Control Node Figures 6d and 6e show the call set-up process in network configurations in which an ISTP 232 serves as a call control node, according to the invention. As explained above, the EISUP trunks are not required if an ISTP serves as the call control node, according to the invention. The only conditioning network required is the programming of each ISTP enabled to serve as a call control node. Each ISTP serving as the call control node must be programmed to examine each message of the ISUP it processes, to determine whether this ISUP message refers to a service subscribed to by a selected call. For example, if the caller A using the telephone 25 dials the digits 555-4653 and the dialed digits are received or directly through a line appearance of the line 27 or via the PSTN 202, SP 34 consult its tables of translation and determines that the call should be sent forward to SP 36 over the logs of the regular ISUP 206. Therefore, SP 34 reserves an inactive member of regular 200 ISUP logs and formulates or collects an IAM which is sent forward on the links A 206 to the ISTP 232. This ISTP 232 examines the IAM and determines that it relates to a call selected according to the invention. The ISTP 232, therefore, creates a call control table entry indicating that a selected call request has been received, but otherwise passes the IAM transparently over the links A 206 to SP 36. Upon receiving the IAM , SP 36 consults its guidance tables and determines that it serves the called number which may be of the dialed digits or it may be a converted number changed by the ISTP 232 using the translation table of the dialed number, as described before. SP 36 maps the IAM parameters into an ISDN call setup message, which is sent forward to IP 214. This IP 214 returns a message preceding the ISDN call and an alert and connection message . The IP 214 also prepares and dispatches a data message on the data link 218 to the data network 226, which forwards the data message on the data link 228 to the ISTP 232 to supply the ISTP 232 with the IP address. of this IP 214, which may not be available on the ISTP 232. Upon receiving from IP 214 the ISDN call that proceeds and alerts and connects the messages in SP, this SP 36 formulates and dispatches the messages of the ISUP, ACM and ANM , to SP 34, which sends them over the A 206 links to the ISTP 232. That ISTP232 examines the messages and writes a record to indicate that the call has been answered. It then transparently carries forward the messages on links A 206 to SP 34, completing the connection to allow interaction between caller A and IP 214, as will be described in more detail below. Figure 6 shows a similar call set-up sequence in which the ISTP 232 serves as the call control node, according to the invention, but the IP 214 serves by a switch in the PSTN 202. The whole message of the ISUP proceeds through the ISTP 232 as described above. The only difference in the call set-up sequence is that when the SP 36 queries its guidance tables upon receipt of the SP 34 IAM, it determines that the called number is served by an SP anywhere in the PSTN. Therefore, it collects a modified AMI appropriately and issues it in the PSTN that is routed to the SP that serves IP 214. Next, the call setup proceeds as described above.

Sequences in Call Means As explained above, one purpose of the invention is to allow a portion of the call to be rejected without the disconnection of the calling party. The following examples describe the call sequences in which the called party is disconnected and the call is redirected to another called party. It will be understood by those skilled in the art that the invention allows a situation in which the calling party is disconnected and the call is redirected under the control of the called party to a new termination. Figures 7a-7c illustrate three sequences in the middle of the call, which follow the call setup sequences illustrated in Figures 6a-6e. In this example, the interaction is with an IP 214. As will be understood by those skilled in the art, the interaction can also take place with a person in a call center or the like. In any case, the results are substantially the same. In this example, during the interaction between caller A and IP 214, caller A, for example, enters selections that indicate to IP 214 that caller A wants to speak to a sales representative of the subscriber to the service. In accordance with the prior art processes, IP 214 would typically include a front end switch, which guides the call back out into the PSTN to the sales representative and forms a bridge with the calling party to the new termination. , thus occupying many of the duplicate facilities and wasting resources from the PSTN. According to the invention, upon receiving the request of the caller A, during the interaction which results after the call establishment, the IP 214 instructs the call control mode of the caller A request and the call is redirected from the node of call control, allowing a more efficient use of the resources of the PSTN. This sequence can be repeated any number of times so that, for example, if the caller A decides to enter into conversation with sales to acquire a membership, the sales can send a message through the data network 226 to the control node of calls to reconnect caller A with accounting to arrange financing, or similar. Likewise, after the financing has been arranged, accounting can again instruct the call control node to connect the caller A to a program administrator who establishes the initiation and orientation sessions. All of the above is handled transparently and without inconvenience to caller A while it requires a minimum of entry from the service subscriber.

VSP to IP Figure 7a shows an exemplary sequence to the call mita, where the VSP 208 serves as the call control node and the selected calls are routed to the VSP 208 using a set of addresses associated with the trunks 212 of the EISUP . Following the call setup, as described above with respect to Figures 6a and 6b, the P 214 determines by analyzing the DMTF tones, for example, that the user wishes to speak to a sales representative of an IP 214 subscriber, for example. therefore, it formulates a data message the lime dispatches on the line 218 of data connected to the data network 226. This data network 226 sends forward the data message on the data connection 228 to the VSP 208. The message contains a serial number of the call identification, as described above, which the VSP 208 uses to identify the call in the call control table. Upon receiving the message, the VSP 208 locates the call in the call control table and uses the data in the table as an ISUP REL message, which dispatches -a. SP 36 for releasing the call connected to the terminating end of EISUP 2112 in SP 36 in CIC 18. This causes SP 36 to generate an RLC message, which returns to VSP 208. This VSP 208 captures the message and queries the table signaling control of the common channel, which indicates that the message o must be roasted back to SP34, since it is not expected and would create an error condition. SP 36 also formulates and sends an ISDN release trunk message to IP 214, which returns a confirmed ISDN release message and the terminating end of the call is thus rejected. The VSP 208 then formulates an IAM message using OPC 001, DPC 002 and CIC 18 derived from the signaling control board of the common channel with the dialed number being the number passed from IP 214 in the field of the reconnected number of the message of TCP / IP 300 (see Figure 4b). Therefore, the call setup continues in Figures 6a or 6b at the "INS" point and the call set-up proceeds until the conversion results between the calling party and a sales representative for the service subscriber. It will be understood by those skilled in the art that unless the sales representative connects to SP 36 or PSTN 202 via an ISDN line, the calling process can not be identical to the sequence shown in Figures 6a and 6b, since the sales representative can have a regular line termination to SP 36 or can be connected to a PBX or similar. However, the call setup is otherwise identical to that shown in Figures 6a and 6b, except that the ACM and ANM messages are not passed back to SP 34, because SP 34 would not expect such messages.

VSP to IP Using the Back Loop Trunks of the ISUP Figure 7b shows the mid-call tracking sequence for an embodiment of the invention, using the back loop trunks of the ISUP. As before, caller A is supposed to indicate to IP 214 that it wants to talk to a sales representative of the service subscriber. The IP 214, therefore, formulates a data message, which dispatches the data link 218 to the data network 226. This data network 226 sends forward the message on the data link 228 to the VSP 208. Upon receiving the message, the VSP 208 determines by examining the reconnection number field of the message (Figure 4b) that must regulate the call to a new termination. . The VSP 208, therefore, queries its Call Control Table and determines that the call is associated with the caller A. The VSP 208 returns a confirmation that the instructions are received and instructs the IP 214 to release resources. In order to release the terminating end of the call, the VSP 208 compiles an ISUP REL message having an OPC of 003 and a DPC of 000, derived from the common channel signaling control table, as described above. . In this example, in order to simplify CIC, the numbering of the VSP 208 is assigned to two point codes 001 and 003 to allow the exit and entry ends of the rear loop trunks of the ISUP to have the same CIC code. Therefore, the REL message compiled by the VSP 208, which has the OPC of 003 and the DPC of 000, includes a CIC of 2, which indicates the entry end of member 1 of the back azo trunk group of the ISUP. Upon receipt of the REL message, SSP A releases its capture from CIC 2 in the ISUP 234 trunk group and compiles a message from the RLC that has an OP of 000 and a DPC of 003, which returns to VSP 208 In order to reject the balance of the call, SP 34 then resends a message from the trunk of the ISDN releasing this ISDN to IP 214, which returns in response a confirmed message of ISDN release to SP 34 The VSP 208 then compiles an IAM message using the look-up tables described above. The IAM has an OPC of 003, a DPC of 000 and a CIC of 2 with a calling number, which is a CLID of caller A and the called number, which is the number of the sales representative of the service subscriber provided. to VSP 208 by IP 214 in the reconnection number field, as described above. The IAM is broadcast on a common channel signaling network, where it is sent to SP 34 via STP pair 204 and the establishment of a connection to the called party is carried out in much the same way as shown in the " INS "of Figure 6c, with the exception that the called party may not necessarily subscribe to the ISDN, as explained above. Again, ACN and ANM messages are passed back from VSP 208 to SP 34.

ISTP to IP Figure 7c shows a half-calling sequence for the call set-up sequences shown in Figures 6d and 6e, where the call control node is an ISTP 232. In this case, the IP 214 at receiving an indication of the caller A, compiles a data message which it transmits on the data link 218 to the data network 226. This data network 226 sends forward the data message on the data link 228 to the ISTP 232. Upon receipt of the data message, the ISTP 232 consults its call control table and determines that the data message will release the call in question. caller progress A. The ISTP 232, therefore, uses the lookup tables of the call control node to formulate an ISUP REL message that has an OPC of 000, a DPC of 002 and a CIC of 14. The message The REL is transmitted over the A 206 links to the SP 36, which responds with the ISUP RLC message that has an OPC of 002, a DPC of 000 and a CIC of 14. The ISTP 232 does not forward the RLC over SP 34 , as it would be during the normal call process. Rather, it discards the message as instructed by an entry in the common channel signaling control table. The SP 36 also dispatches a trunk message of the ISDN to release this ISDN to the IP 214 and receives in response a confirmed message of release of the IP 214. The ISTP 232 again uses the query tables of the call control node for formulate an IAM message that has an OP of 000, a DPC of 002, a CIC of 14, a CLID calling number from the call control table and a dialed number of the reconnection number supplied by IP 214. Transfers that message about links A 206 to SP 36 that initiates the call setup to the sales representative of the service subscriber by much with the same sequence as shown in Figures 6d and 6e, beginning at the "INS" symbol. In the same way, as explained above, the ACM and the ANM are not passed back through ISTP 214 to SP 34. As explained above, the subscriber's sales representative may have a connection to the Internet or to some other network 226 of data, which allows the release of your end of the call, when the caller A requires a further service that can better be provided by another member of your organization or some other organization. After satisfying all the needs of the caller A, the call goes to the final release sequence shown, for example, in Figure 8. This final release sequence can be initiated by any of at least three initiators: a release command received of IP 214 in response to an input from caller A; a disconnection initiated by the caller A that hangs; or, on the one hand to which the call was transferred by sending a data release message to the call control or hanging node. In Figure 8, the call control node illustrated is a VSP 208. It will be understood by those skilled in the art that the same release sequence applies to the ISTP 232.

In the sequence illustrated in Figure 8, caller A is supposed to indicate to IP that they will probably disconnect rather than make the transfer to sales, for example. Upon receipt of the indication, the IP 214 formulates a data message, which dispatches over the data connection 218 to the data network 226. This data network 226 sends forward the message on the data link 228 to the VSP 208. Upon receiving the disconnection request, the VSP 208 formulates a RE message having a DPC of 000 and an OPC of 001, with a CIC of the call circuit which extracts from the common channel signaling control board and dispatches the message to SP 34. It then formulates a release message having an OPC of 001, a DPC of 002 and the CIC of the call circuit derived from the same table and forward the message to SP 36. Upon receiving the REL message, SP 34 releases the trunk member of the EISUP indicated by the CIC and applies the dial tone to line 27 of the part. This line 27 of the part hangs up and SP 34 sends an RLC message that has an OPC of 000 and a DPC of 001, with a CIC of the call circuit back to VSP 208. SP 36 returns a similar message after receiving an ISDN disconnect signal from IP 214. Then it returns an ISDN release acknowledgment to IP 214, which releases all resources. Similar termination release sequences are used for all call control nodes and all release primers, as will be understood by those skilled in the art.

Considerations of the Complex Network As will be appreciated, the network configurations described above are simple configurations in which the trunk groups of the EISUP are provided between only a pair of switches. In most embodiments, there may be situations where the trunks of a EISUP are provided between a plurality of adjacent switches. Those skilled in the art will understand that this complicates making the decision at the call control node, because the particular OPC will not necessarily indicate the captured EISUP trunk for any particular call. In terms of the simplicity of the provision and conditioning of the network, the exclusive use of the posterior loop trunk groups is the least complicated implementation scheme. If trunk groups of EISUP are used, there are, however, provisioning methods that allow reliable and efficient use of the invention within the PSTN. For example, where several groups of trunks of the EISUP end up in a single switch, the CIC codes for each member of each trunks group must be assigned only and those unique assignments must be carried through the terminating end of each EISUP. While there is a consideration of provision, it does not present a problem. It will also be understood by those skilled in the art that, under certain circumstances, it may be advantageous to designate a given switch as a Point of Presence (POP) in the network for the guidance of selected calls, in accordance with the invention, in order to simplify the conditioning and provision.

INDUSTRIAL APPLICABILITY The widening of telecommunications services, currently offered to the public and the introduction of competition in many telecommunication markets, has driven an increasing pressure on the corresponding service providers to offer a wider range of less expensive customer services. and more innovative. The accepted approach to special services in the telephone network switched to the public has been carried out by carrying out the Intelligent Advanced Network call model, which depends on the software activators (program) integrated in the network switching nodes and the access to remote databases, to provide route information for enhanced call services.

With the current familiarity and acceptance of the Internet, however, telephone service customers are entitled to offer more flexible services than those enabled by the AIN call model. The invention provides a method and apparatus for enabling the offering of novel services that integrate the functionality of the Internet into the PSTN and allow customers to control their calls from within the switching network. The methods and apparatus, in accordance with the invention, are useful in a wide range of applications, including novel customer services, novel business services, and improved call center architectures, including distributed call center architectures. The invention allows unprecedented control of network switching points, while adhering to the fundamental protocol, to allow novel call control, while ensuring the stability of the network.

Claims (45)

R E I V I N D I C A C I O N S

1. A method for dynamically directing selected calls through an intelligent switched telephone network, using common channel signaling messages, generated by a call control node in a common channel signaling control path for the call, which includes the steps of directing a call through the network to a first termination, in response to a number dialed by a calling party, so that the common channel signaling path passes through the call control node, CHARACTERIZED because : the call control node includes a common channel signaling interface, to receive and send common channel signaling messages related to the selected calls directed there, by sets of guides and link sets, associated with the selected calls to the node control of calls; elements for examining common channel signaling messages and transparently forwarding selected common channel signaling messages to an adjacent signaling node; elements for generating common channel signaling messages, in response to predetermined criteria; elements for guiding individual calls virtually switched through them; and, a data interface for receiving data inputs from a data network, not associated with the common channel signaling network; and the call control node assumes control of the selected calls, and determines a new termination for each selected call, this new termination being indicated, directly or indirectly, by previously defined criteria, and initiating a release of the first termination, which uses at least one common channel signaling message, without releasing the calling party; and initiating redirection of the call to the new termination, using at least one other common channel signaling message.

2. A method, as claimed in claim 1, in which the data entries are messages that guide the calls.

3. A method, as claimed in claim 1, in which the data entries are messages that control the calls.

4. A method, as claimed in claim 2, in which the call control node is a virtual switching point which also comprises an element for the translation of the dialed numbers in the call address information.

5. A method, as claimed in claim 4, in which the element for translation is enabled to translate the dialed digits in the call address information, based on a value of at least one previously defined variable.

6. A method, as claimed in claim 7, in which at least one previously defined variable includes the time of the day and the day of the week.

7. A method, as claimed in claim 1, wherein the call control node is a Point of Intelligent Signal Transfer (ISTP).

8. A method, as claimed in claim 1, wherein the call control node is a virtual switching point and the selected calls are directed so that the common channel signaling control path passes through the switching point , by: designating at least one trunk (connection of lines) as a group of trunks, to which the selected calls are directed in at least one network switching node; creating address table entries in at least one switching node to direct the selected calls to at least one group of logs; and create the sets of links and sets of addresses, so common channel signaling messages, related to calls guided to at least one group of trunks, are directed to the virtual switching point.

9. A method, as claimed in claim 8, in which this at least one trunk is a back loop trunk, having an exit end and an entry end, respectively connected to a trunk side of the network switching node .

10. A method, as claimed in claim 9, wherein the output end and the input end are circuit identification codes in sequence, assigned, respectively.

11. A method, as claimed in claim 8, wherein this at least one trunk is an internal switching trunk.

12. A method, as claimed in any of the preceding claims, wherein the selected calls are directed based on the dialed digits.

13. A method, as claimed in claim 12, in which the dialed digits are translated to a Service Control Point (SCP), before the address occurs.

14. A method, as claimed in any of claims 1 to 13, wherein the selected calls are guided based on the identification of the calling line.

15. A method, as claimed in any of claims 1 to 13, wherein the selected calls are guided based on the identification of the calling line and the digits, marked.

16. A method, as claimed in any of the preceding claims, in which the call control node passes only the selected common channel signaling messages back to an originating end of a call, until this call is disconnected.

17. A method, as claimed in claim 16, wherein the common channel signaling messages are the messages of the Signaling System 7 (SS7) and the User Part of the Public Switched Telephone Network (ISUP).

18. A method, as claimed in claim 16, wherein the selected common channel signaling messages comprise the Full Release (REL), Release (RLC), Response (ANS) and Full Address (ACM) messages.

19. An apparatus for dynamically directing selected calls through an intelligent switched telephone network, comprising a common channel signaling interface, for receiving messages from, and sending messages through, a common channel signaling network, a memory, for store at least one of the messages a common channel signaling network,. a memory for storing programs, a processor, for examining messages a common channel signaling network, received at the interface a common channel signaling network and for generating messages a common channel signaling network for controlling call connections, CHARACTERIZED because: the apparatus further includes elements for forwarding, transparently, messages a selected common channel signaling network to an adjacent signaling node; elements for generating the messages a common channel signaling network, in response to predetermined criteria; elements for guiding individual calls commutated virtually through them and, a connection to a data network not associated with the network a common channel signaling network, for receiving data inputs to control the selected calls; and the programs enable the processor to guide the calls routed virtually through the device, and to evaluate the previously defined criteria, to determine an action regarding the control of the call connections.

20. An apparatus, as claimed in claim 19, in which an Intelligent Peripheral is also connected to the data network and the apparatus and Intelligent Peripheral exchange messages on the data network.

21. An apparatus, as claimed in claims 19 or 20, in which a call center is also connected to the data network and the apparatus and the call center exchange messages over the data network.

22. An apparatus, as claimed in any of claims 19-21, in which the apparatus is assigned with a service switching point code in the network and this apparatus appears in the network as a Switching Point (SP).

23. An apparatus, as claimed in any of claims 19-22, in which at least some of the forward sent messages are modified before they are reissued in the common channel signaling network.

24. An apparatus, as claimed in claim 23, in which the messages that are modified are modified only to an extent that the Point of Origin Code and the Destination Point Code are changed to appear logically correct in a node, which is a receiver of the message.

25. A system for dynamically directing selected calls through an intelligent switched telephone network, whereby calls are routed through a first termination for the call, so that a signaling control path for the call passes through the node call control, CHARACTERIZED because: the call control node is a virtual node in the network switching plane and a physical node in a common channel signaling plane of the network and includes elements to examine the messages of common channel signaling and forward, transparently, some of the selected common channel signaling messages to an adjacent signaling node; elements for generating common channel signaling messages in response to predetermined criteria; elements for guiding individual calls, virtually switched through them; and, a connection to a data network or associated with the common channel signaling network, to receive data inputs to control the selected calls; and the call control node is enabled to release calls in a forward direction from a call control node and reconnect the calls to a new termination, specified by the data from an external source, received via the connection to the data network, without releasing a calling party, associated with the call.

26. A system, as claimed in claim 25, wherein the element for establishing the first termination of the call comprises: at least one trunk, in at least one group of trunks, designated to direct selected calls and a set of associated addresses, which guide signaling messages of the common channel, associated with this at least one trunk to the call control mode.

27. A system, as claimed in claims 25 or 26, in which this at least one trunk is a back trunk of a loop, connected to the side of the trunk of a switching node in the network.

28. A system, as claimed in claims 25 or 26, in which this at least one trunk is a trunk of the ISUP between switches in the network.

29. A system, as claimed in any of claims 25 to 28, wherein the data network is the Internet .

30. A system, as claimed in claim 25, in which the data is supplied by a called party at the call termination.

31. A system, as claimed in claim 30, in which the data is supplied by the calling party, who acts internally with a machine connected to the data network.

32. A system, as claimed in claim 31, wherein the calling party acts internally with the machine through the call connection.

33. A system, as claimed in any of claims 30 or 31, wherein the calling party acts internally with the machine, using the DTMF.

34. A system, as claimed in any of claims 30 or 31, wherein the calling party acts internally with the machine, using command voices.

35. A system, as claimed in any of claims 25 to 34, wherein the element for releasing the call in a forward direction from the call control node comprises common channel signaling messages, generated by the control node of calls.

36. A system, as claimed in claim 35, in which the common channel signaling messages are the messages of the User Part of the ISDN of the SS7.

37. A system, as claimed in any of claims 25 to 36, wherein the call control node is a Virtual Switching Point (VSP), which is assigned with a Switching Node Point Code.

38. A system, as claimed in any of claims 25 to 36, wherein the call control node is a Smart Signaling Transfer Point (ISTP).

39. A system, as claimed in claim 37, in which the VSP is enabled to execute queries to a Service Control Point (SCP), if the ported number requires translation.

40. A system, as claimed in claim 38, in which the ISTP is enabled to execute queries to a Service Control Point, if a ported number requires translation.

41. A system, as claimed in claim 39, in which the VSP performs queries to the SCP, using messages from the Application Part of the Transaction Capabilities of the SS7 (TCAP).

42. A system, as claimed in claim 40, in which the ISTP performs queries to the SCP, using messages from the Application Part of the Transaction Capabilities of the SS7 (TCAP).

43. A system, as claimed in claim 39, in which the VSP uses a data network to query the SCP.

44. A system, as claimed in claim 40, in which the ISTP uses a data network to query the SCP.

45. A system, as claimed in claim 37, in which the VSP compiles the call records in the billing records, for billing the selected calls to a service subscriber.