An enhanced IP telephone taking advantage of the many call-related functions provided by existing PBX systems. The enhanced IP telephone includes a thin computer telephony integration (CTI) client transmitting to a CTI server a command associated with a particular call-related function provided by a PBX unit. The CTI server translates the command to a PBX command, and transmits the PBX command to the PBX unit for providing the particular call-related function to the enhanced IP telephone. Instead of the thin CTI client, the enhanced IP telephone may include an enhanced IP signaling protocol stack for accessing the PBX functionality using its signaling protocol. A CTI translator translates the enhanced signaling command to a CTI command for transmitting to the CTI server.
Architecture For Transporting Pbx Signaling Codes Via Sip
An integrated PBX and SIP telephony communications network supporting transmission of private signaling codes over SIP. A telephone emits a private signaling code for accessing a particular PBX function. An enhanced IP translator coupled to the telephone encapsulates the private signaling code within a SIP invite, request, response, information, notification message, and/or instant message. The SIP message with the encapsulated private signaling codes is transmitted over a communications network to a PBX. The PBX is also coupled to an enhanced IP translator that receives the SIP message and decapsulates the embedded private signaling codes. The codes are transmitted to the PBX for rendering a function for the telephone. A SIP user agent aware of the private signaling codes may also access functions provided by the PBX by encapsulating/decapsulating private signaling codes into/from SIP messages.
A switch with emulation client in a data communication network adhering to a Session Initiation Protocol (SIP). The switch is coupled to a legacy telephone which initiates or receives telephonic communication with a SIP-observant telephone or another legacy telephone over the SIP-observant network. The emulation client translates between PBX and SIP messages to allow the legacy telephone to communicate seamlessly over the SIP observant network. The switch is further coupled to a location server which storing redirection information for redirecting calls made to a particular number. The calls are preferably redirected based on the location where the user is scheduled to be at the time of the call.
An enhanced IP telephone taking advantage of the many call-related functions provided by existing PBX systems. The enhanced IP telephone includes a thin computer telephony integration (CTI) client transmitting to a CTI server a command associated with a particular call-related function provided by a PBX unit. The CTI server translates the command to a PBX command, and transmits the PBX command to the PBX unit for providing the particular call-related function to the enhanced IP telephone. Instead of the thin CTI client, the enhanced IP telephone may include an enhanced IP signaling protocol stack for accessing the PBX functionality using its signaling protocol. A CTI translator translates the enhanced signaling command to a CTI command for transmitting to the CTI server.
Methods And Devices For Achieving Parallel Operation Between Ip And Analog Phones
Michael S. Wengrovitz - Concord MA, US Andrew Nelson - Earlwood, AU
Assignee:
Alcatel-Lucent USA Inc. - Murray Hill NJ
International Classification:
H04L 12/66
US Classification:
370352, 370356, 37920201, 709205
Abstract:
An IP bridge phone for operatively coupling one or more analog phones to a digital communications network to support parallel communications is disclosed. The IP bridge phone includes a phone handset interface adapted to connect to a phone handset, an analog phone interface, and a network interface. A mixing module is adapted to receive a first voice signal from the handset interface, a second voice signal from the analog phone, and a third voice signal from the data communications network; generate a plurality of combined signals from the first voice signal, the second voice signal, and the third voice signal; and transmit each of the plurality of combined signals to one of the plurality of interfaces. The plurality of combined signals provide parallel communications between the phone handset, the analog phone, and the remote calling party.
Timucin Ozugur - Garland TX, US Michael S. Wengrovitz - Concord MA, US
Assignee:
Alcatel Lucent - Paris
International Classification:
H04L 12/56
US Classification:
370352, 4554121, 463 42, 709206
Abstract:
A dynamic contact list management (DCLM) method and system operable in a communications network environment. In one embodiment, upon receiving a communication request from an originating party over a first network towards a networked party, a query is generated for determining and locating an IM address associated with the originating party. Upon establishing presence awareness between the originating party and the networked party, the IM address of the originating party is added to a contact list database associated with the networked party. The IM address, which is persistent at least for the duration of a first communication session established responsive to the communication request, may be used for engaging in a second communication session between the parties.
Session Initiation Protocol Routing Using Voice Cookies
A system and method for establishing a SIP session between calling and called end-points. A SIP server receives a SIP INVITE request for establishing the SIP session. The SIP INVITE request includes standard routing information in its header. The SIP server receives the SIP INVITE request and responds by transmitting to the calling end-point a HTML link associated with a web server. In rendering the contents of the HTML link, the calling end-point transmits an HTTP request to the web server including voice cookie information. The voice cookie information may include information gathered about the caller such as the caller profile information, transaction information, caller intent information, or recent and past history associated with a web site domain. The web server forwards the voice cookies to the SIP server. The SIP server determines an address of the called end-point as a function of the routing information in the SIP INVITE request and the voice cookie information.
Client-Based Integration Of Pbx And Messaging Systems
This invention integrates instant messaging, presence, and other collaborative capabilities with conventional PBX functionality through use of a PBX-Messaging Integration Client (PMIC). The invention in its several embodiments features a method and system for using the PMIC-based computer interface to perform off-hook/on-hook presence notification for a PBX phone, establish media sessions concurrent with PBX telephonic communication, execute custom call treatment in conjunction with a PBX phone, implement call transfer capability between a PBX phone and numerous other devices, and provide PBX call control. The PBX is generally enabled with computer telephony integration (CTI) and, depending on the embodiment, a Voice-over-Internet Protocol (VoIP) such as Session Initiation Protocol (SIP). The invention empowers enterprise workers with a diverse, unified and integrated set of both PBX functions and SIP-based collaboration tools.
Mutualink since Sep 2012
Senior Director
Alcatel-Lucent 1999 - Aug 2012
Distinguished Member of Technical Staff
Bose Corporation 1994 - 1999
Manager Professional Audio Electronics group
Atlantic Aerospace Electronics Corporation 1986 - 1994
Director Acoustic Systems
Education:
Massachusetts Institute of Technology 1981 - 1986
ScD, EE
Skills:
Integration Telecommunications Ip Sip Product Management Wireless Voip Unified Communications Ethernet Mobile Devices Cloud Computing Network Design Switches Lte Mpls Tcp/Ip Qos Internet Protocol Wireless Technologies Voice Over Ip Routing Ss7 Fiber Optics Dwdm Network Engineering Unix Wimax Cdma Gsm Iptv Rf Session Initiation Protocol 3G Sdh Wifi 4G Snmp Rtsp
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