WO2009085608
"10. A method as recited in claim 1 wherein said limiting comprises establishment of a three-way call（通話）."
"20. A method as recited in claim 1 wherein said event comprises a call origination event（発呼イベント）.
21. A method as recited in claim 1 wherein said event comprises a call termination event（着呼イベント）."
"27. A method as recited in claim 1, further comprising classifying the event as one of a mobile- originated（移動体発信）(MO) or mobile-terminated（移動体着信）(MT) voice or short message service (SMS) call; determining that the mobile device is within said area of interest（対象エリア）using the low accuracy location function; determining that the mobile device is potentially（可能性がある）in the quiet zone; invoking（呼び出す）the high accuracy location function before a call setup operation is performed; determining that the mobile device is not within the quiet zone; permitting the call setup operation to be completed and thus allowing the mobile device to begin the call（呼を開始）; periodically invoking a mid-call（呼最中）location function; determining by the mid-call location function that the mobile device has moved into the quiet zone; and terminating（終了）the call."
WO2009061727
"1. A method (800) to be executed at least in part in a computing device for facilitating communications in an integrated telephony system (100, 200) through dual forking（二重分岐）, the integrated telephony system (100, 200) comprising a first system and a second system with at least one of the systems being connected to an external communications network, the method comprising: receiving (802) a request for a phone call（通話のリクエスト）from a user provisioned for dual forking in the integrated telephony system (100, 200); determining a source identifier, a destination address（宛先アドレス）, and a preferred identity for the user; if the destination address is outside the integrated system (100, 200): routing（経路指定）the request to the second system after replacing the source identifier with an external identity derived from the preferred identity and the destination address with a destination phone number, wherein the external identity identifies the user on either one of the first and second system in a transparent manner to a called party（呼び出されるパーティ）at the destination phone number; and upon establishing connection with the called party at the destination phone number, facilitating communications through an end device registered to the requesting user in one of the first system and the second system; if the destination address is inside the integrated system: determining the called party by performing a reverse number lookup（逆引き検索）based on the destination address; determining an available end device for the called party in at least one of the first system and the second system; and facilitating communications through the available end device for the called party in one of the first system and the second system.
2. The method (800) of claim 1, further comprising: receiving an inbound call（着呼）from an external caller addressed to（宛てられた）an external identity assigned to one of the users of the integrated system (100, 200) provisioned for dual forking; determining an identifier for the called user based on the called external identity, wherein the external identity is the same on both systems for the called user; routing (806) a call request based on the inbound call to available end devices registered to the called user in the first and second systems using the identifier; upon establishing connection with the called user through one available end device, facilitating communications with the external caller（発呼者）and cancelling (808) call requests to other end devices registered to the called user（呼び出されるユーザ；＊着呼ユーザ？）."
"5. The method of claim 4, further comprising: preventing an endless loop of calls within the integrated system (100, 200) by employing one of: not routing a call that has been received from one of the first and the second systems to the same system; rejecting a suspected loop call（ループ呼）in a destination system of the first and second systems and returning a warning response to an origination system（発信元システム）of the first and second systems; and inserting a parameter to a call request in the destination system for notifying the origination system to suppress routing of a call back to the destination system."
US2006045248
"1. A method of identifying arbitrage, comprising:
determining whether originating（発信）and terminating（着信）call detail records (CDRs) are correlated and obtaining correlated candidate pairs from the determined CDRs;
establishing（確かめる）whether a correlated candidate pair of the obtained correlated candidate pairs is a unique pair; and
if established that a correlated candidate pair is unique, determining an amount of arbitrage based on the unique correlated candidate pair.
2. The method of claim 1, wherein the originating CDRs are determined by obtaining a CDR and determining whether a call is leaving（通話が発信されている）a monitored network（監視対象ネットワーク）and whether a called party jurisdictional state（着呼側の管轄権）is in the monitored network.
"15. The method of claim 14, wherein the CDR fields of the unique correlated candidate pair include trunk types, billing jurisdiction, calling number（発呼側番号）, charge number and routing carrier."
WO2000014935
"1. In an Internet Protocol telephone system using a telephone（電話）to place and receive voice（電話通話を発呼および受信）over Internet Protocol (VolP)-based and public switched telephone network (PSTN)-based telephone calls, a method comprising:
detecting an off-hook condition from the telephone;
receiving a sequence of signals generated by the telephone; buffering at least a first signal generated by the telephone;
attempting to detect a predetermined signal that signifies a VoIP-based call（ＶｏＩＰ方式の通話）; and
intercepting subsequent signals in the sequence, absent（除いて）the at least first signal that was buffered, and placing the VoIP-based call（通話を発呼する）via an internet when the predetermined signal is detected.
2. The method according to claim 1 further comprising:
detecting a presence of an incoming telephone call（着呼）; and
signaling a user of the telephone during the VoIP-based call of the presence of the incoming telephone call.
3. The method according to claim 2 wherein the user of the telephone is signaled via an audible tone.
4. The method according to claim 3 wherein the audible tone is of a first ringing cadence（呼出音）when the incoming telephone call is a VoIP-based call and the audible tone is of a second ringing cadence when the incoming telephone call is a PSTN-based call"
"6. In an Internet Protocol telephone system using a digital wireless handset to place and receive internet-based telephone calls using a voice over Internet Protocol (VoIP) as well as public switched telephone network (PSTN) based calls via a common compression decompression engine, a method comprising:
converting analog signals to digital signals at the digital wireless handset to be transmitted to a network premises gateway when placing an outgoing call（発信通話を発呼する）;
translating digital signals to a format compatible for a network used in completing the outgoing call at the network premises gateway, wherein the network is a PSTN for PSTN-based calls and an internet for VoIP-based calls; and
converting digital signals transmitted from the network premises gateway to analog signals at the digital wireless handset when receiving an incoming call（着呼を受信する）."
US2011066707
"8. The method of claim 7, wherein the first user is at least one of an originator and target of the communication request and wherein the second identity is an enterprise identity of the at least one of the originator（発信者）and target（着信者）."
US2005180547
"1. A method of identifying a caller（発信者）of a call from the caller to a recipient（着信者）, the method comprising:
(a) receiving a voice input from the caller;
(b) applying characteristics of the voice input to a plurality of acoustic models, which comprises a generic acoustic model and acoustic models of any previously identified callers, to obtain a plurality of respective acoustic scores;
(c) identifying the caller as one of the previously identified callers or a new caller based on the plurality of acoustic scores; and
(d) if the caller is identified as a new caller in step (c), generating a new acoustic model for the new caller, which is specific to the new caller."
