WO2009039576A1

WO2009039576A1 - System, apparatus and method for communication

Info

Publication number: WO2009039576A1
Application number: PCT/AU2008/001426
Authority: WO
Inventors: Antonio Cantoni; Hooi Lit Ng; John Frank Siliquini; Kent Gibson; Kevin Alston Fynn; Stephan Bettermann; Steven Antony Ivandich
Original assignee: Thebuzz Corp Pty Ltd
Priority date: 2007-09-25
Filing date: 2008-09-25
Publication date: 2009-04-02

Abstract

Communication is established between a source telephony device (2) and a destination telephony device (24) over a telecommunications network by providing an interface (14) to couple a private branch telephone exchange (4), and the telecommunications network. The private branch telephone exchange (4) services a plurality of telephony devices (2) via branch telephone lines (5). The telecommunications network provides at least two telephone lines (1) to the private branch telephone exchange (4). An available telecommunications channel is accessed on one of the telephone lines (1), That telephone line (1) and a branch telephone line (5) of the private branch telephone exchange (4) are bridged by the interface (14). Communication is routed between the source telephony device (2) and the destination telephony device (24).

Description

"System, Apparatus and Method for Communication"

Field of the Invention

This invention relates to a system, apparatus and method for establishing telephone calls between telephony devices, for example, using a combination of Public Switched Telephone Network and Voice over Internet Protocol technologies.

Throughout the specification unless the context requires otherwise, the word "comprise" or variations such as "comprises" or "comprising", will be understood to imply the inclusion of a stated integer or group of integers but not the exclusion of any other integer or group of integers.

Throughout the specification unless the context requires otherwise, the word "include" or variations such as "includes" or "including", will be understood to imply the inclusion of a stated integer or group of integers but not the exclusion of any other integer or group of integers.

Background Art

The following discussion is intended to facilitate an understanding of the present invention. However, the discussion is not an acknowledgement or admission that any of the material referred to was published, known or part of the common general knowledge of the person skilled in the art in any jurisdiction as at the priority date of the application.

Circuit switching technology used within the Public Switched Telephone Network ("PSTN") is the most commonly used networking technology to offer and support end-to-end voice communication services. Voice over Internet Protocol ("VoIP") is an alternative technology for offering voice communication services with at least one advantage related to the reduced costs associated with using Internet Protocol network technology compared to the PSTN. However, the prerequisites for a user to take advantage of the cost benefits of VoIP, typically includes (i) Internet Protocol access connection at the user premises via an Internet Service Provider, (ii) specialised hardware and software, and (iii) a VoIP service provider. For many users, whether they are residential or business users, it may not be viable or possible to acquire these prerequisites in order to use VoIP technology. Calling card technology is one method whereby this can be achieved but this technology requires the user to perform many steps, for example, dialling one number to access the service, then typing in a personal identification number ("PIN"), then typing in the required destination number.

Disclosure of the Invention

In accordance with one aspect of the present invention, there is provided a method for establishing communication between a source telephony device and a destination telephony device over a telecommunications network, the method comprising:

providing an interface between a private branch telephony device, servicing a plurality of source telephony devices, the said interface interfacing with the private branch telephony device via branch telephone lines, and a telecommunications network providing at least two telephone lines for the private branch telephony device,

accessing an available telecommunications channel on one of said at least two telephone lines via the interface,

bridging the one telephone line and a branch telephone line of the private branch telephony device, and

routing communication between the source telephony device and the destination telephony device.

Preferably, the method further comprises detecting an off-hook condition on the said branch telephone line of the private branch telephony device in response to the initiation of a telephone call at the source telephony device. Preferably, the method further comprises capturing call data inputted at the source telephony device to initiate the telephone call.

Preferably, the method further comprises transmitting the call data inputted at the source telephony device to a call server using the available telecommunications channel to thereby establish a telecommunications call leg prior to bridging the said one telephone line and the said branch telephone line of the private branch telephony device.

Preferably, routing communication between the source telephony device and the destination telephony device comprises establishing a destination call leg from the call server to the destination telephony device using the call data transmitted on the available telecommunications channel.

Preferably, routing communication between the source telephony device and the destination telephony device further comprises bridging the telecommunications call leg and the destination call leg.

In one alternative, the method comprises, in addition to using the available telecommunications channel to transmit the call data inputted at the source telephony device, transmitting the call data to the call server on the telecommunications channel via at least one media gateway.

In a first alternative, accessing an available telecommunications channel on one of said at least two telephone lines comprises accessing an existing free telecommunications channel.

In a second alternative, accessing an available telecommunications channel on one of said at least two telephone lines comprises establishing a new free telecommunications channel.

Preferably, the method further comprises providing the private branch telephony device as a private branch telephone exchange.

Preferably, the method further comprises detecting an on-hook condition on said branch telephone line in response to termination of the telephone call, and terminating the destination call leg to terminate communication between the source telephony device and the destination telephony device.

In accordance with another aspect of the present invention, there is provided a telecommunications system for establishing communication between a source telephony device and a destination telephony device over a telecommunications network, the telecommunications system comprising:

an interface to couple a private branch telephony device, servicing a plurality of source telephony devices, the said interface interfacing with the private branch telephony device via branch telephone lines, with the telecommunications network which provides at least two telephone lines for the private branch telephony device, and

a call server to communicate with the interface, wherein

the interface is operable to access an available telecommunications channel on one of said at least two telephone lines and to bridge the one telephone line and a branch telephone line of the private branch telephony device, and

the call server is operable to route communication between the source telephony device and the destination telephony device.

Preferably, the interface is operable to detect an off-hook condition on the said branch telephone line of the private branch telephony device in response to the initiation of a telephone call at the source telephony device.

Preferably, the interface is operable to capture call data inputted at the source telephony device to initiate the telephone call.

Preferably, the interface is operable to transmit the call data inputted at the source telephony device to the call server using the available telecommunications channel to thereby establish a telecommunications call leg, prior to bridging the said one telephone line and the said branch telephone line of the private branch telephony device. Preferably, the call server is operable to establish a destination call leg from the call server to the destination telephony device using the call data transmitted on the available communications channel.

Preferably, the call server is operable to bridge the telecommunications call leg and the destination call leg.

Preferably, the call server is operable to receive call data on the telecommunications channel via at least one media gateway.

Preferably, the interface is operable to access an available telecommunications channel on one of said at least two telephone lines by accessing an existing free telecommunications channel.

Preferably, the interface is operable to access an available telecommunications channel on one of said at least two telephone lines by establishing a new free telecommunications channel.

Preferably, the private branch telephony device comprises a private branch telephone exchange.

In accordance with a further aspect of the present invention, there is provided an interface for a telecommunications system for establishing communication between a source telephony device and a destination telephony device over a telecommunications network, wherein

the interface is operable to:

be coupled between a private branch telephony device, servicing a plurality of source telephony devices, the said interface interfacing with the private branch telephony device via branch telephone lines, and the telecommunications network which provides at least two telephone lines for the private branch telephony device, access an available telecommunications channel on one of said at least two telephone lines, and

bridge the one telephone line and a branch telephone line of the private branch telephony device.

Preferably, the interface is operable to transmit the call data inputted at the source telephony device to a call server using the available telecommunications channel to thereby establish a telecommunications call leg, prior to bridging the said one telephone line and the said branch telephone line of the private branch telephony device.

The present invention provides a method, system and apparatus whereby users that are not necessarily directly connected to the Internet (and possibly connected via the PSTN only), may more simply, reliably and with reduced steps, use, for example, VoIP technology in order to reduce call tariffs than would otherwise have been the case if the PSTN were used directly. The present invention may additionally (or alternatively) be used to provide a telephony service that is of higher quality than existing VoIP services and/or provide a telephony service that has superior reliability features than existing VoIP services.

Brief Description of the Drawings

The present invention will now be described, by way of example only, with reference to the accompanying drawings, in which:

Figure 1 schematically illustrates an example of an existing telephone communication system arrangement where a Private Branch Exchange (PBX) is connected to four PSTN lines;

Figure 2 schematically illustrates the architecture and system components of a telephone communication system in accordance with an aspect of the present invention

Figure 3 schematically illustrates an embodiment of a configuration of the PTID apparatus, employed in the present invention, interfacing between the exiting PBX source telephony device and the PSTN shown in Figure 1 ; and

Figure 4 schematically illustrates an embodiment of the resulting end-to-end communications between the source and destination telephony devices resulting from the use of the present invention.

Best Mode(s) for Carrying Out the Invention

Throughout the specification and claims, unless the context requires otherwise, the word "comprise" or variations such as "comprises" or "comprising", will be understood to imply the inclusion of a stated integer or group of integers but not the exclusion of any other integer or group of integers.

The term 'telephony device' is used herein to describe any device or apparatus that communicates within telephony environments. Examples include fixed landline phones, Private Branch (telephony) Exchanges (PBX), cellular radio phones, cordless phones, internet phones or any other such devices. The term 'PSTN line' is used herein to describe any type of analogue line or Integrated Services Digital Network (ISDN) channel that is used to make telephone calls via the PSTN. The term 'telephone service provider' is used herein to describe any service provider of telephone termination services and includes traditional PSTN telecommunication companies and also includes VoIP service operators or providers.

Many Small and Medium size Businesses (SMB) utilize a number of PSTN line connections to support telephony services amongst its employee staff. An example of an existing arrangement that makes use of four PSTN lines and a Private Branch Exchange (PBX) is depicted in Figure 1. In the arrangement shown in Figure 1 , the four PSTN lines 1 provide the (source) telephony devices 2 (which may be in the form of telephones) with telephone access to the PSTN 3. The telephones 2 are connected to the PBX 4 via extension telephone lines 5. PBX 4 controls the access of the telephones 2 to the PSTN 3.

The present invention may be used in the types of environment depicted in Figure 1.

Figure 2 illustrates an embodiment of the architecture and system components used in the present invention. The telephone communication system 10 illustrated in Figure 2 comprises a call server 12, a multiline PSTN Telephony Interface Device (hereon referred to as PTID) 14, a branch telephony device which may be provided as a PBX system 4 and two media gateways 18 and 20. It will be appreciated from the following description that other numbers of call servers12, PTIDs 14, PBXs 4 and media gateways 18 and 20 may be used, and the present invention is not limited to the number of these components illustrated in Figure2.

The system components, their functionality and their inter-working will now be described.

A PTID 14 is hardware or software, or a combination of hardware and software components, that interfaces between the source telephony device (for example a PBX 4,) and the PSTN 3, so as to operatively couple the PBX 4 and the PSTN 3. This is shown in Figure 3. The interfacing of the PTID 14 with the PBX 4, or branch telephony device, and the PSTN 3 may be by any suitable means. For example, the PTID 14 may interface with the branch telephony device (PBX 4) via branch telephone lines (PBX lines 15).

The server 12 and the PTID 14 respectively comprise functional modules to perform the functions described herein.

The PTID 14 is used in the establishing and maintaining call connections initiated by the telephony devices. The PTID 14 can establish and maintain multiple call connections for the telephony devices at the same time. This means that the present invention allows multiple calls on the telephony devices to be made simultaneously to multiple and unique call destinations.

The PTID 14 interfaces to the source telephony devices 2, via the PBX 4. The PTID 14 is functional to detect, for example, the DTMF tones transmitted by a source telephony device 2, and to record in memory information inputted by a user of the source telephony device 2. Usually, the information inputted by the user at a source telephony device 2 relates to the phone number of the destination telephony device 24 with which the user intends to establish a telephone communication.

The PTID 14 also interfaces to the PSTN 3 and is functional to establish regular telephone communication channels through the PSTN 3 to other telephony devices. The PTID 14 is also functional to assist in the establishment of communications channel(s) between itself and the call server 12. This communications channel will be referred to as the PTID/Call Server Communications Channel ("PCSCC").

The PTID 14 is able to transmit and receive call information, via the PCSCC, to and from the call server 12. The PTID 14 is able to transmit and receive such call information, for example, via the transmission or reception of DTMF tones or via modem data communications. When making a call, this is achieved by transmitting from the PTID 14 to the call server 12 the details of the phone number of the destination telephony device 24 with which the user, at the source telephony device 2, requires to establish a telephony communication. The call server 12 uses this information and assists in establishing telephone communications between the source telephony device 2 and a destination telephony device 24.

In addition to supporting signalling between the PTID 14 and the call server 12, the PCSCC is also used to support the end-to-end telephone communications between the source telephony device 2 and destination telephony devices 24.

An existing PCSCC is referred to as an active PCSCC if it is currently supporting an end-to-end communication between a source telephony device

2 and a destination telephony device 24. An existing PCSCC is referred to as a free PCSCC, or a PCSCC that is available, if it is currently not supporting an end-to-end communication between a source telephony device 2 and a destination telephony device 24. The PTID 14 may also have operative features that detect calls to and from local devices, i.e. other telephony devices, including a branch telephone or source telephone, in order to allow the source telephony device 2 to operate as it normally would.

The call server 12, under control of software, co-ordinates the establishment of telephone communications between telephony devices. The call server 12 maintains a database which contains information regarding current call costs associated with multiple telephone service providers (including VoIP service providers). The call server 12 also maintains a database relating to the quality and/or availability of the voice service being offered by telephone service providers (for example, VoIP service providers). This information is used to establish optimum service operating behaviour, for example, by choosing which telephone service provider to use for establishing call legs so as to minimise call costs or by choosing which telephone service provider to use for establishing call legs so as to maximise call quality. The call server 12 can also keep databases storing additional information such as customer preferences related to calls that allow it to make decisions about setting up calls, for example, a preference to minimise call costs or a preference to maximise call quality.

The call server 12 is also functional to assist in the establishment of PCSCCs with the PTID 14. The call server 12 is also functional to receive call details from the PTID 14, such as the phone number of the destination telephony device 24, via an established PCSCC or from the PTID 14 via the Internet. After receiving call details from the PTID 14 (such as the phone number of the source telephony device 2), it is the function of the call server 12 to facilitate the end-to-end telephone communications establishment between the source telephony device 2 and the destination telephony device 24. This can proceed in a number of ways, depending on the preference of the user of this system. For example, the call could proceed at least in the following way: In one case, the call server 12 initiates a call between itself and the subsequently acquired destination telephony device 24 using a specified telephone service provider. The call server 12 is then functional to "bridge" the call leg 26 to the PTID 14 (using the previously established PCSCC) with the call leg 28 established to the destination telephony device 24, thereby creating end-to-end telephony communications between the source telephony device 2 and destination telephony device 24. The "bridging" of call legs 26 and 28 in this way (between the source telephony device 2 and the destination telephony device24) enables the voice call between source and destination to be perceived as being direct.

The call server 12, under control of software, is operable to establish or tear down call legs as instructed by the PTID 14. Phone calls made by the call server 12 can be via circuit switched network (e.g. the Public Switched Telephone Network (PSTN) 3) or they can be via the Internet (e.g. using Voice over Internet Protocol (VoIP) technology) or a combination of these and other technologies. The structure and operation of the Internet and the use of communication protocols such as Internet Protocol (IP), Session Initiation Protocol (SIP), H.323, etc for establishing VoIP communications is well known and need not be further described herein. If VoIP technology is used to establish call legs, then in order to couple PSTN calls to VoIP (and vice versa) the media gateways 18 and 20, illustrated in Figure 2, are used. The use of media gateways 18 and 20 for such coupling is well known and, as such, need not be described in any further detail herein.

The call server 12 can also include the functions associated with user authorisation, user current country location information, generation of user call detail records for billing, etc. There need be only one call server 12 for the system to operate. However, there may be more than one call server 12 used for the purposes of: reducing the signalling delay and/or increasing scalability and/or for redundancy purposes. For example, a call server 12 could be used for establishing PCSCCs while other call servers 12 could be used for establishing call legs. Each call server 12 is callable via a set of G_k (k is an integer greater than zero) telephone numbers NCS₉ (g = 1 , 2, ... G_k), for example, through the use of Direct Inward Dialling (DID) numbers or 1-300 or 1 -800 telephone services or a combination of these.

PTID Interfaces

The PTID system supports the following interfaces:

N PSTN lines 1. The individual lines 1 are referenced as the f^h PSTN line where 1 ≤i ≤N. The PSTN lines 1 interface to the PSTN exchange 3.

• N PBX lines 15. The individual lines 5 are referenced as the f^h PBX line where 1 ≤i ≤N. The PBX lines 15 interface to the PBX 4 and the PTID 14 acts as the PSTN exchange for the PBX 4.

In practice, when a PBX 4 is used, there will be at least two lines, i.e. N ≥2.

Operation

1. Outgoing calls

One embodiment of a method for establishing communication between a source telephony device 2 and a destination telephony device 24 over a telecommunications network, to thereby make a call between the source telephony device 2 and the remote or destination telephony device 24, using the communications system of the present invention, will now be described. The destination telephony device 24 may be, for example, a fixed landline phone or a mobile (i.e. cellular) phone. It is assumed that the user of the source telephony device 2 has a PTID 14 suitably interfaced between the source telephony device 2 (typically via a PBX 4) and the PSTN 3.

Step 1. To make a telephone call, the user at the source telephony device 2 lifts the handset (or equivalent action) and dials the destination phone number in the usual way. The PBX 4 detects this condition by detection of the off-hook condition of the source telephony device and proceeds to go off-hook on one of the free PBX lines 15 (say the f^h line). The PTID 14 is operable to detect the off-hook condition on the

/" PBX line 15. Proceed to Step 2.

Step 2. The PTID 14 captures the call data inputted by the user, i.e. the DTMF digits dialled by the user on the I^th PBX line 15, and records into memory the destination telephony device 24 phone number, N_d dialled by the user at the source telephony device 2. During the dialling process, the PTID 14 will recognise if the number being dialled is not a bypass number. Bypass numbers may include emergency (e.g. 000, 911 , etc) and other numbers such as, for example, free-call and local charge rate numbers (which may commence with digits such as 13x and 1800 ). If the number N_d dialled is not a bypass number, the procedure continues from Step 3 below. If the number N_d dialled is detected as a bypass number, then the PTID 14 seizes an unused /'⁷ PSTN line 1 , transmits on that f PSTN line 1 the DTMF tones representing the digits already dialled by the user, then bridges the f^h PBX line 15 and the/" PSTN line 1.

Step 3. If there is an existing free PCSCC (channel) on some /c"⁷ PSTN line 1 , then the PTID 4 transfers the number N_d being dialled to the call server 12 via DTMF tones on that PCSCC. This expedites the transfer of the number N_d being dialled to the call server 12. If there is no existing free PCSCC (channel) then the PTID 4 initiates the process of establishing a new free PCSCC, i.e. a new available PCSCC to the call server 12. The establishment of new PCSCCs can be carried out, for example, in a way described in the section later herein titled Establishing a PCSCC . Once a free PCSCC is established (as described, for example in the section, Establishing a PCSCC) for this call on some k?^h PSTN line 1 , the PTID 14 transfers the number N_d being dialled to the call server 12 via DTMF tones on the /c"¹ PSTN line 1. It may also be possible to pass the number N_d being dialled to the call server via other means, such as using the Internet and the Internet Protocol. Proceed to Step 4.

Step 4. Once the transfer of the number N_d to the call server 12 is completed, the PTID 14 bridges the k"¹ PSTN line 1 with the /" PBX line 15 thereby connecting the user to the lt^b PSTN line 1. This step facilitates the end-to-end telephone communications establishment between the source telephony device and the destination telephony device 24.

Proceed to Step 5.

Step 5. When the call server 12 receives the number N_d from the PTID 14 on a free PCSCC, or otherwise, the call server 12, based on the value of the number N_d and any user profile information, will nominate a telephone service provider and proceed to establish the call to the telephone number N_d of the destination telephony device 24 using the selected telephone service provider. This call establishes the destination call leg 28. The Call Server then bridges the PCSCC call leg 26 (which terminates on the U^h PSTN line 1 at the PTID 14) with the destination call leg 28. In this way the source telephony device 2 and destination telephony device 24 perceive the call to be direct. The resulting end-to-end communications between the source telephony device 2 and the destination telephony device24 is depicted in Figure 4. Proceed to Step 6. Step 6. When the on-hook is detected on the f^h PBX line 15 by the PTID 14, the PTID 14 transmits a DTMF sequence on the associated active PCSCC on the fd^h PSTN line 1 that allows the call server to terminate the destination call leg 24 and generate a call detail record (CDR) for this call. The DTMF sequence transmitted by the PTID 14 on the relevant active PCSCC could, for example, be either the "BBB*" or the "CCC^*" DTMF sequences. The PCSCC channel at this stage, however, may or may not be terminated and could be used for subsequent new calls initiated by a source telephony device 2. The procedure for keeping the PCSCC open or alternatively terminating this channel is described in the section Semi-Permanent PCSCC.

2. 3^rd Party Incoming Calls - No Call Server Call Back is Pending

If the PTID 14 detects an incoming call (when not expecting a call back call from the call server 12) on a /^Λ PSTN line 1 , a free U^h PBX line 15 is bridged to the f^h PSTN line 1 , thereby transferring ringing and any Call ID (CID) information to the PBX 4. The incoming call is then processed by the PBX 4 in the usual way.

Establishing a PCSCC

The following procedures and methods may be used for establishing a new PCSCC: METHOD 1 :

Step L At the PTID 14:

The PTID 14 dials the call server 12 using NCS₉ on a seized f PSTN line 1 thereby requesting the call server 12 to call back the PTID 14 to facilitate the establishment of a PCSCC. Proceed to Step 2.

Step 2. At the Call Server 12:

Upon detecting an incoming call from the PTID 14 on NCS₃, the call server 12 records the Caller ID (CID) of the source telephony device 2 into N₈ and signals "busy" (i.e. does not answer the call) and terminates the call from the PTID 14. If N_s corresponds to an authorised user then the call server 12, based on the value of N_s and any user profile information, will nominate a telephone service provider that will use to establish the PCSCC to the initiating PTID 14. The call server 12 will then call back the PTID 14 by making a call to the telephone number N₃ of the source telephony device 2 using the selected telephone service provider. This facilitates the establishment of the PCSCC. Proceed to

Step 3.

Step 3. At the PTID 14:

The PTID 14 that initiated the establishment of the PCSCC waits for "busy" or timeout on the/" PSTN line 1. After this, the PTID 14 then goes on-hook on the f^h PSTN line 1 and waits for call server call back.

Proceed to Step VERIFY.

METHOD 2:

Step L At the PTID 14:

The PTID 14 notifies the call server 12 to initiate the establishment of a call back to the PTID 14 to facilitate the establishment of a PCSCC via the Internet and the Internet Protocol, whereby the PTID 14 signals via the Internet to the call server 12 the value of N₃, N₃ being the telephone number of the PTID (i.e. the source telephony device telephone number) requesting the call back. Proceed to Step 2.

Step 2. At the Call Server 12:

If N_s that was received from the PTID 14 via the Internet corresponds to an authorised user, then the call server 12, based on the value of N₃ and any user profile information, will nominate a telephone service provider that will use to establish the PCSCC to the initiating PTID 14. The call server 12 will then call back the PTID 14 by making a call to the telephone number N₃ of the source telephony device using the selected telephone service provider. This facilitates the establishment of the PCSCC. Proceed to Step 3.

Step 3. At the PTID: The PTID 14 that initiated the establishment of the PCSCC waits for call server 12 call back. Proceed to Step VERIFY.

METHOD 1 and METHOD 2:

Step VERIFY: At the PTID 14:

The call server 12 call back can occur on any k?^h PSTN line 1 at the

PTID 14. The call server 12 call back call is detected and verified via the Caller ID (CID) of the call server 12 appearing on the /c"¹ PSTN line 1 OR, if no CID is available or CID detection is not possible, then it is assumed, immediately after requesting (either by the use of METHOD 1 or METHOD 2) a call back from the call server 12, that the next incoming call on any /c* PSTN line 1 is the call back call from the call server 12. In the latter case, the procedure for verifying the next incoming call is the call back from the call server 12 is as follows:

a. The PTID 14 answers the next incoming call on the /c"⁷ PSTN line 1 assuming it is the call server 12 call back call. b. Once the /c"⁷ PSTN line 1 is answered, the PTID 14 waits for a predetermined DTMF identification sequence from the call server 12 (for example, the DTMF sequence "BBB") c. IF no call server 12 identification DTMF sequence is detected THEN IF there is a free PBX line 15 (the m^th PBX line) THEN

- Seize the m^th PBX line 15 by providing a ringing signal on the m'" PBX line 15,

- Give "false" ring call progress tone to incoming PSTN caller on the /c"⁷ PSTN line 1 ,

- On off-hook detection on m^th PBX line 15, bridge the \t^h

PSTN line 1 with the m^th PBX line 15,

ELSE - bridge the /c"⁷ PSTN line 1 with the /* PBX line 15 (the f PBX line 15 is the PBX line 15 as specified in the section previously herein titled Outgoing Calls).

END IF

Wait for call server 12 call back call as the next incoming call starting at a. above

ELSE

New free PCSCC verified on the /c"⁷ PSTN line 1

END IF

Semi-permanent PCSCC

Keeping a PCSCC semi-permanently "open" is advantageous for reducing the total call setup time by eliminating the need to establish new free PCSCCs every time a new call is initiated from a user at the source telephony device 2. The feature could operate as follows:

1. To Keep the PCSCC Open for Subsequent Use at the PTID 14 for New

Calls:

On the termination of a call through an active PCSCC, the PTID 14 signals the call server to keep the currently established PCSCC to remain open (and subsequently free). The signalling message from the PTID 14 to the call server 12 for keeping the PCSCC open could be, for example, the DTMF sequence "BBB^*". Upon recept of this signalling message from the PTID 14, the call server 12 will not terminate the PCSCC and the PCSCC becomes free and will be available by the call server 12 for receiving new values of the number A/^ from the PTID 14 and then subsequently establishing new end-to- end communications as described in the section previously herein titled Outgoing Calls.

2. To terminate the PCSCC: On the termination of a call through an active PCSCC or on any free PCSCC₁ the PTID 14 signals the call server 12 to terminate the currently established PCSCC. The signalling message from the PTID 14 to the call server 12 for terminating the PCSCC could be, for example, the DTMF sequence "CCC*". Upon recept of this signalling message the call server 12 terminates the PCSCC.

Additionally, algorithms can be implemented at the PTID 14 or at the call server 12 to ensure that there is a maximum amount of time that the PCSCCs are permitted to remain open. Once that permitted maximum amount of time elapses, the PCSCCs will be terminated if they are not active. Further, algorithms can be developed to ensure that there is a maximum number of PSTN lines 1 that are permitted to be used to support free PCSCCs. For example, the following algorithm could be used to ensure that a maximum of M (M ≤N) PCSCCs remain free at any one time and that individual PCSCCs do not remain open for more that T seconds:

IF any free PCSCC has been free for greater than T seconds THEN

PTID 14 signals the call server 12 to terminate this PCSCC

END IF

On call termination event on the /c"⁷ PSTN line 1 supporting an active PCSCC₁

IF time since first establishing PCSCQ is greater than 7 seconds THEN

PTID 14 signals call server 12 to terminate this PCSCQ

ELSE

IF the number of currently free PCSCCs is less than M THEN

PTID 14 signals call server 12 to keep this PCSCQ open

ELSE

PTID 14 signals call server 12 to terminate this PCSCQ END IF

END IF

Also, it may be advantageous to always allow a minimum number of unused PSTN lines 1 to exist for 3^rd party incoming calls. For example, the following algorithm could be used at the PTID 14 to ensure that a minimum of V (V ≤N) free PSTN lines 1 are left free for the reception of incoming calls:

On detection of new established 3^rd party incoming call OR on a call termination event on the /c"¹ PSTN line 1 supporting an active PCSCC

IF number of free PSTN lines 1 is less than VTHEN

PTID 14 signals call server 12 to terminate the oldest existing free PCSCC

END IF.

Modifications and variations such as would be apparent to a skilled addressee are deemed to be within the scope of the present invention.

Claims

1. A method for establishing communication between a source telephony device and a destination telephony device over a telecommunications network, characterised in that it comprises:

providing an interface between a private branch telephony device, the private branch telephony device servicing a plurality of source telephony devices, the said interface interfacing with the private branch telephony device via branch telephone lines, and a telecommunications network providing at least two telephone lines for the private branch telephony device,

accessing an available telecommunications channel on one of said at least two telephone lines,

2. A method according to claim 1 , characterised in that it further comprises detecting an off-hook condition on the said branch telephone line of the private branch telephony device in response to the initiation of a telephone call at the source telephony device.

3. A method according to claim 1 or 2, characterised in that it further comprises capturing call data inputted at the source telephony device to initiate the telephone call.

4. A method according to any one of the preceding claims, characterised in that it further comprises transmitting the call data inputted at the source telephony device to a call server using the available telecommunications channel to thereby establish a telecommunications call leg, prior to bridging the said one telephone line and the said branch telephone line of the private branch telephony device.

5. A method according to claim 4, characterised in that routing communication between the source telephony device and the destination telephony device comprises establishing a destination call leg from the call server to the destination telephony device using the call data transmitted on the available telecommunications channel.

6. A method according to claim 5, characterised in that routing communication between the source telephony device and the destination telephony device comprises bridging the telecommunications call leg and the destination call leg.

7. A method according to any one of claims 4 to 6, characterised in that, in addition to using the available telecommunications channel to transmit the call data inputted at the source telephony device, the method comprises transmitting the call data to the call server on the telecommunications channel via at least one media gateway.

8. A method according to any one of the preceding claims, characterised in that accessing an available telecommunications channel on one of said at least two telephone lines comprises accessing an existing free telecommunications channel.

9. A method according to any one of claims 1 to 7, characterised in that accessing an available telecommunications channel on one of said at least two telephone lines comprises establishing a new free telecommunications channel.

10. A method according to any one of the preceding claims, characterised in that it further comprises providing the private branch telephony device as a private branch telephone exchange.

1 1. A method according to any one of claims 5 to 10, characterised in that it further comprises detecting an on-hook condition on said branch telephone line in response to termination of the telephone call, and terminating the destination call leg to terminate communication between the source telephony device and the destination telephony device.

12. A telecommunications system for establishing communication between a source telephony device and a destination telephony device over a telecommunications network, the telecommunications system comprising:

an interface to couple a private branch telephony device, servicing a plurality of source telephony devices, the said interface interfacing with the private branch telephony device via branch telephone lines, and the telecommunications network which provides at least two telephone lines for the private branch telephony device, and

a call server to communicate with the interface, wherein

13. A telecommunications system according to claim 12, characterised in that the interface is operable to detect an off-hook condition on the said branch telephone line of the private branch telephony device in response to the initiation of a telephone call at the source telephony device.

14. A telecommunications system according to claim 13, characterised in that the interface is operable to capture call data inputted at the source telephony device to initiate the telephone call.

15. A telecommunications system according to claim 14, characterised in that the interface is operable to transmit the call data inputted at the source telephony device to the call server using the available telecommunications channel to thereby establish a telecommunications call leg, prior to bridging the said one telephone line and the said branch telephone line of the private branch telephony device.

16. A telecommunications system according to claim 15, characterised in that the call server is operable to establish a destination call leg from the call server to the destination telephony device using the call data transmitted on the available communications channel.

17. A telecommunications system according to claim 16, characterised in that the call server is operable to bridge the telecommunications call leg and the destination call leg.

18. A telecommunications system according to any one of claims 12 to 17, characterised in that the call server is operable to receive call data on the telecommunications channel via at least one media gateway.

19. A telecommunications system according to any one of claims 12 to 18, characterised in that the interface is operable to access an available telecommunications channel on one of said at least two telephone lines by accessing an existing free telecommunications channel.

20. A telecommunications system according to any one of claims 12 to 18, characterised in that the interface is operable to access an available telecommunications channel on one of said at least two telephone lines by establishing a new free telecommunications channel.

21. A telecommunications system according to any one of claims 12 to 18, characterised in that the private branch telephony device comprises a private branch telephone exchange.

22. An interface for a telecommunications system for establishing communication between a source telephony device and a destination telephony device over a telecommunications network, wherein

the interface is operable to:

be coupled between a private branch telephony device, servicing a plurality of source telephony devices, the said interface interfacing with the private branch telephony device via branch telephone lines, and the telecommunications network which provides at least two telephone lines for the private branch telephony device,

access an available telecommunications channel on one of said at least two telephone lines, and

23.An interface according to claim 22, characterised in that the interface is operable to detect an off-hook condition on the said branch telephone line of the private branch telephony device in response to the initiation of a telephone call at the source telephony device.

24.An interface according to claim 23, characterised in that the interface is operable to capture call data inputted at the source telephony device to initiate the telephone call.

25.An interface according to claim 24, characterised in that the interface is operable to transmit the call data inputted at the source telephony device to a call server using the available telecommunications channel to thereby establish a telecommunications call leg, prior to bridging the said one telephone line and the said branch telephone line of the private branch telephony device.

26.An interface according to any one of claims 22 to 25, characterised in that the interface is operable to access an available telecommunications channel on one of said at least two telephone lines by accessing an existing free telecommunications channel.

27.An interface according to any one of claims 22 to 25, characterised in that the interface is operable to access an available telecommunications channel on one of said at least two telephone lines by establishing a new free telecommunications channel.