摘要:
Methods and apparatus for implementing communication services such as voice dialing services are described. In one Centrex based voice dialing embodiment, voice dialing service subscribers are given access to personal voice dialing records including calling entries via the Internet as well as via telephone connections. Each calling entry normally includes the name and, optionally nickname, of a party to be called. It also includes one or more telephone numbers associated with each name. Different telephone number identifies, e.g. locations, can be associated with different names. A user can create or update entries in a voice dialing directory using text conveyed over the Internet or speech supplied via a telephone connection. In order to facilitate updating and maintenance of voice dialing directories over the Internet speaker independent (SI) speech recognition models are used. When a calling entry is created via the Internet the text of the name is processed to generate a corresponding speech recognition model there from. When an entry is created via speech obtained over the telephone, a speech recognition model is generated from the speech and a text name is generated is generated using speech to text technology. To avoid having to hang-up and initiate a new voice dialing call the outcome of a voice dialing call is monitored and the subscriber is provided the opportunity to initiate another call using voice dialing if the first call did not complete successfully e.g., goes unanswered.
摘要:
Methods and apparatus for implementing communication services such as voice dialing services are described. In one Centrex based voice dialing embodiment, voice dialing service subscribers are given access to personal voice dialing records including calling entries via the Internet as well as via telephone connections. Each calling entry normally includes the name and, optionally nickname, of a party to be called. It also includes one or more telephone numbers associated with each name. Different telephone number identifies, e.g. locations, can be associated with different names. A user can create or update entries in a voice dialing directory using text conveyed over the Internet or speech supplied via a telephone connection. In order to facilitate updating and maintenance of voice dialing directories over the Internet speaker independent (SI) speech recognition models are used. When a calling entry is created via the Internet the text of the name is processed to generate a corresponding speech recognition model there from. When an entry is created via speech obtained over the telephone, a speech recognition model is generated from the speech and a text name is generated is generated using speech to text technology. To avoid having to hang-up and initiate a new voice dialing call the outcome of a voice dialing call is monitored and the subscriber is provided the opportunity to initiate another call using voice dialing if the first call did not complete successfully e.g., goes unanswered.
摘要:
Methods and apparatus for implementing communication services such as voice dialing services are described. In one Centrex based voice dialing embodiment, voice dialing service subscribers are given access to personal voice dialing records including calling entries via the Internet as well as via telephone connections. Each calling entry normally includes the name and, optionally nickname, of a party to be called. It also includes one or more telephone numbers associated with each name. Different telephone number identifies, e.g. locations, can be associated with different names. A user can create or update entries in a voice dialing directory using text conveyed over the Internet or speech supplied via a telephone connection. In order to facilitate updating and maintenance of voice dialing directories over the Internet speaker independent (SI) speech recognition models are used. When a calling entry is created via the Internet the text of the name is processed to generate a corresponding speech recognition model there from. When an entry is created via speech obtained over the telephone, a speech recognition model is generated from the speech and a text name is generated is generated using speech to text technology. To avoid having to hang-up and initiate a new voice dialing call the outcome of a voice dialing call is monitored and the subscriber is provided the opportunity to initiate another call using voice dialing if the first call did not complete successfully e.g., goes unanswered.
摘要:
Methods and apparatus for implementing communication services such as voice dialing services are described. In one Centrex based voice dialing embodiment, voice dialing service subscribers are given access to personal voice dialing records including calling entries via the Internet as well as via telephone connections. Each calling entry normally includes the name and, optionally nickname, of a party to be called. It also includes one or more telephone numbers associated with each name. Different telephone number identifies, e.g. locations, can be associated with different names. A user can create or update entries in a voice dialing directory using text conveyed over the Internet or speech supplied via a telephone connection. In order to facilitate updating and maintenance of voice dialing directories over the Internet speaker independent (SI) speech recognition models are used. When a calling entry is created via the Internet the text of the name is processed to generate a corresponding speech recognition model there from. When an entry is created via speech obtained over the telephone, a speech recognition model is generated from the speech and a text name is generated is generated using speech to text technology. To avoid having to hang-up and initiate a new voice dialing call the outcome of a voice dialing call is monitored and the subscriber is provided the opportunity to initiate another call using voice dialing if the first call did not complete successfully e.g., goes unanswered.
摘要:
Communications methods for allowing groups of individuals, such as family members, to contact one another when no one is available to answer calls to a primary, e.g., family, telephone number are described. Calls which go unanswered are connected to a telephony device capable of storing messages and initiating conference calls. If the calling party is determined to be a family member the telephony device accesses a table of contact information corresponding to the unanswered telephone number and offers the caller the opportunity to hear messages from other family members and/or initiate a conference call to one or more family members who have supplied a contact telephone number. Family members can access and update the table of messages and contact number information by telephone or the Internet. Table information can be accessed and updated remotely thereby providing a method of communicating with family members even when no one is at home.
摘要:
AIN based call routing, transfer and conferencing methods and apparatus are disclosed. In various embodiments initial call routing is based on the availability of a party to service a call as determined from a computer system associated with the party selected to service the call. The party's computer system supports a telephone application programming interface (TAPI) which allows a telephone network server to determine the availability of the party selected to service the call from, in part, TAPI obtained telephone line status information. The network server supplies call related data to the computer system of the party assigned to service the call. Call transfer and conferencing operations along with the transfer of call related data are also supported. AIN mid-call triggers are used in some embodiments to support call transfer and conferencing operations.
摘要:
AIN based call routing, transfer and conferencing methods and apparatus are disclosed. In various embodiments initial call routing is based on the availability of a party to service a call as determined from a computer system associated with the party selected to service the call. The party's computer system supports a telephone programming application interface (TAPI) which allows a telephone network server to determine the availability of the party selected to service the call from, in part, TAPI obtained telephone line status information. The network server supplies call related data to the computer system of the party assigned to service the call. Call transfer and conferencing operations along with the transfer of call related data are also supported. AIN mid-call triggers are used in some embodiments to support call transfer and conferencing operations.
摘要:
Methods and apparatus for implementing call forwarding services using AIN techniques and next event list messages are described. Methods of notifying a subscriber of a call forwarded using AIN techniques are also described. In accordance with one feature of the present invention, a subscriber is allowed to set the number of rings which are allowed to occur prior to a call being forwarded. The number of rings, e.g., the ring count, is stored as part of a customer's call processing record (CPR) which is used by a service control point (SCP) to implement the call forwarding service. The ring count information may be updated via the Internet or via a dial-up telephone connection. Call forwarding customers are provided with a distinctive ring, e.g., a ring shorter than an ordinary telephone ring, to notify them when a call is being forwarded. Distinctive rings may be used to distinguish between different forwarding services. For example a different ring may be used for forwarding to voice mail than for selective call forwarding or follow-me call forwarding. The distinctive ring may be implemented by using an SCP to instruct a telephone switch to provide any one of a plurality of different rings which are supported by the telephone switch or by simply causing the subscribers phone to ring for different periods of time which are shorter than a normal ring.
摘要:
Methods and apparatus for billing communications services through the use of AIN functionality are described. Billing information is communicated to a service control point by placing one or two calls to a service subscriber to be billed for a service. The calls encounter a trigger set at a switch. In response to trigger activation the switch sends a message to the service control point. The message includes billing information placed into a billing data field of the call. The service control point extracts the billing information received in the message or messages from the switch and bills the called service subscriber based on the received billing information. In the single billing call embodiment the device initiating the billing call is responsible for determining the amount of time for which the service is to be billed, e.g., conference duration, in addition to other billing information.
摘要:
AIN based methods and apparatus for transitioning telephone numbers and customers from the PSTN to a VOIP network are described. AIN line number portability features are used to allow a few gateway switches that interconnect the PSTN and VOIP networks to service customers whose telephone numbers were originally serviced by several remote PSTN switches. AIN LNP triggers are used to forward PSTN calls, directed to the PSTN switch previously used to service a telephone number, to the gateway switch assigned to route such calls to the IP network. AIN triggers set at the gateway switch insure that the subscriber with the ported telephone number continues to receive AIN services provided before the telephone number was ported to the IP network. Calls from ported telephone numbers to telephones in the PSTN are billed from PSTN switches through the use of AIN functionality and triggers set at the gateway switch.