摘要:
The present invention relates to establishing a communication channel between two communication systems having computer telephony integration (CTI). Many CTI systems are configured into incoming and outgoing lines according to the anticipated demands. Consequently, if sufficient outgoing or incoming capacity is unavailable at a given time to support a communication channel of required characteristics, the users of such a system must conventionally wait until sufficient capacity becomes available. However, the present invention determines who the intended addressee is and can instruct the CTI system associated with that addressee to instigate the establishment of a communication channel to the user who originally desired the connection. The instructions can be sent to the intended addressee using for example, another communication network, such as the Internet or other data communications network. This arrangement also allows the call to be placed in the reverse direction, where this is favourable for tariff reasons. The invention also includes the automatic selection of a telephone network carrier based on tariff data requested over a data network such as the Internet.
摘要:
A voice processing system is connected to a switch via multiple telephone lines, and provides a set of line objects, each line object being associated with one of the physical telephone lines. The line object allows a demarcation to be made between the underlying voice processing system software, and external business applications. Thus a line object supports a set of methods such as Get DTMF Tone, Play Audio, Answer Call, and End Call, to allow the external business applications to perform desired operations on a telephone line. These methods are invoked via a set of corresponding IVR action objects, which in turn are integrated into the business application. The business application itself, and its IVR actions, regard the line objects effectively as servers to provide IVR functionality. The business application may therefore run partially or completely on a separate physical machine from the IVR system itself.
摘要:
A voice processing system comprises a computer workstation 80 running a voice system software and a telephony interface module 70 which is attached via a trunk line 100 to a telephone switch 10. The computer system 80 and the telephony interface module 70 are connected by a standard data connection 230 such as a SCSI connection over which voice data is exchanged. The SCSI interface is provided with a voice device driver capable of handling voice data by means of an appropriate set of commands.
摘要:
A method of supporting Voice over Internet Protocol (VOIP) media codecs on a voice processing system stores voice response segments in each of a plurality network native formats. Each of the network native formats corresponds to a media codec. The method receives a call from a caller over an IP network. The method determines a negotiated codec for the call. If the call requires a required voice response segment, the method retrieves the required voice segment in the network native format corresponding to the negotiated codec. The method then sends the required voice segment in the network native format corresponding to the negotiated codec to the caller over IP network. If the call includes a voice message from the caller, the method stores the voice message from the caller in the network native format corresponding to the negotiated codec. If the call requires retrieval of a voice mail message for the caller, the method retrieves the voice mail message. If the voice mail message is in the network native format corresponding to the negotiated codec, the method sends the voice mail message in the network native format corresponding to the negotiated codec to the caller over the IP network. If the voice mail message is not in the network native format corresponding to the negotiated codec, the method converts the voice mail message into the network native format corresponding to the negotiated codec, and sends the converted voice mail message in the network native format corresponding to the negotiated codec to the caller over the IP network.
摘要:
A first Internet telephone system 620 attempts to call with a second Internet telephone system 630 via the Internet 600. However, the second Internet telephone system 630 is not logged onto the Internet at the time of the call. In response to the failed attempt to call, the first Internet telephone system prompts the user to send voice mail to the user of the second Internet telephone system. This results in a phone call over the Internet between a voice mail system 610 and the first Internet telephone system, allowing a greeting to be heard, and a message to be stored. This message may be subsequently retrieved, either using an Internet telephone system over the Internet, or using a standard phone over the conventional telephone network.
摘要:
The invention relates to a voice processing system capable of varying the speed of output of digitized audio data stored therein. The digitized audio data is stored using blocks of LPC coefficients. Each block is sufficient to allow twenty milliseconds of speech to be generated therefrom. Periodically, or selectably, the utilization of particular blocks is repeated resulting in a decrease in the speed of output of the speech synthesized therefrom. Alternatively, selectably blocks of LPC coefficients are omitted from use thereby producing a corresponding increase in speech output.
摘要:
The disclosure concerns a method of processing Internet telephony messages at gateway computer such as an IBM RISC/6000 system, comprising the steps of: receiving an Internet telephony message in a first compression scheme from a computer running an Internet telephone software application which uses the first compression scheme; converting the Internet telephony message into a compression scheme optionally via an intermediate format; sending the Internet message telephony to another computer running another Internet telephone software application using the second format.
摘要:
A voice server can be located, temporarily allocated, and sent audio. The results are returned to a voice client, and the voice server is deallocated for use by the next person talking into their client browser. Voice channels and IVR ports are initially set up by a switch and the IVR using conventional audio protocols. The voice channels are not initially connected to the client. The switch handles the allocation and deallocation of IVR voice channels without having to communicate further with the IVR. A user indicates to the client device that he wishes to initiate a voice interaction during an X+V session. This translates to a request on the CTRL channel to synchronise XHTML and VXML forms as a trigger for the VXML browser to execute a conversational turn. A multiplexer intercepts this control command and establishes a virtual voice circuit between the client device and an existing open but unattached voice port. The virtual circuit is established without having to set up an RTP channel. The CTRL signal is then forwarded to an interaction manager so that the conversation can take place. At the end of the conversation the virtual circuit is disconnected.
摘要:
Voice data requires large storage resources even when compressed and takes a long time to retrieve. Further the required information cannot normally be directly located and it is difficult to analyze the voice data for statistical information. There is described a method for performing a voice recognition function on a voice telephone conversation to convert the conversation into text data using a voice processing system. The method comprises receiving voice data representing the telephone conversation comprising a first series of speech data from an agent interspersed with a second series of speech data from a client and storing the first and second series of speech data as a single body of voice data for later retrieval. Then a voice recognition function is performed on the voice data to convert it into text data representing the telephone conversation and the text data is stored for later retrieval. Such a solution allows entire days/weeks or even months of conversation to be stored and accessed. Since the memory space required is considerably smaller for text storage it is possible to keep many days of conversation in directly accessible memory that may be searched by a computer. Furthermore it is possible to search for keywords typed from the keyboard and it is not necessary to manually scan the entire conversation for the desired topic.
摘要:
A bus connection controller in a voice processing is for managing the connection of a timeslot on a time-division multiplex (TDM) bus to a port on an adapter. The voice processing system includes basic time-division multiplex (TDM) connection management to enable the coordination of connections between resources such as channels on line cards (SPacks or VPacks), and channels on digital signal processor (DSPs) cards that provide, amongst others things, voice recognition, text-to-speech, fax capabilities and so on. Problems are encountered when a telephone call in a voice processing system ends suddenly because one of the callers hangs up. If the telephony channel has connections with other channels or resources via a TDM bus, callers may hear spurious data. To address this problem each call is associated with its corresponding connection on the TDM bus and each connection is associated with its connection details including the adapters and ports involved in connecting the calls. When one of the calls ends all the relevant ports involved with the connection are immediately disconnected.