The supporting of the managed call recording is used for a call between a calling party and a called party, where a network node receives an invite message that initiates the call, and it is determined that either the calling party or the called party is a managed call recording (MCR) subscriber. A conference bridge is created for the calling party, the called party, and a session initiation protocol recording server (SRS), after the call has been answered by the called party. The network node transmits instructions to play MCR announcements using the conference bridge, prior to the call being connected between the calling party and the called party, where the call is recorded by the SRS.

Enablement of a voice channel being established between an IoT device and a controller through the use of a voice-line service system.

A method, system, and apparatus, including a program encoded on computer-readable medium, for routing communications to a mobile device includes registering a cellular telephone identifier with a visitor location register associated with a bridge mobile switching center adapted to receive communication through a cellular backbone network and route communications over a packet-switched network and determining that cellular communications using the cellular telephone identifier are disabled. A home location register associated with the cellular telephone is notified that the cellular telephone is roaming in a network served by the visitor location register. A communication directed to the cellular telephone is received at the bridge mobile switching center. The received communication is routed to the cellular telephone.

Systems and methods for providing E911 services to a publicly-switched telephone network (PSTN) phone are described. In some embodiments, the systems and methods determine that the PSTN phone is connected to a voice over internet protocol (VoIP) adapter, identify an internet protocol (IP) address for an access point via which the adapter accesses an IP Multimedia System (IMS) network, determine a geographical location associated with the IP address, and enable E911 communications for the PSTN phone over the IMS network via the adapter based on the geographical location.

A system and computer-implemented method for providing telephony communication services for VoIP or analog telephony devices using an on-premises gateway and remotely located VoIP system are described. Some implementations of the methods may include generating a configuration file mapping a plurality of tag extensions to a plurality of analog telephone ports of a high-density analog telephony adapter (HDATA), which may be communicatively coupled to the VoIP system. The VoIP system may transmit the configuration file to the HDATA, receive a telephony service request, and determine a tag extension based on the telephony service request. The VoIP system may transmit an identification of the tag extension to the HDATA and route a telephony service to the HDATA via the communication channel using the tag extension.

Embodiments described herein provide a dynamic voice over Internet Protocol (VoIP) audio quality management mechanism in real time, e.g., when a VoIP call is ongoing. Specifically, when a VoIP call has unsatisfactory audio quality, e.g., due to packet loss, jitter, etc., the dynamic VoIP audio quality management mechanism may redirect the VoIP traffic from the previous endpoint that initiates the VoIP session to a different endpoint within the same carrier. Upon the endpoint redirection, a new call leg is established, allowing re-negotiation or re-configuration of VoIP parameters. The re-negotiated or re-configured VoIP parameters may then be used to conduct the remainder of the VoIP call to improve the audio quality.

A user authentication system that analyzes call forwarding information obtained from telecommunication networks, such as through the use of Signaling System No. 7 (“SS7”) protocols, to detect the possibility of fraud. In response to a request to access a network-accessible service, the system performs an initial authentication of provided user account credentials. The system then obtains a telecommunication subscriber identifier that is associated with the user account. Prior to performing additional device-based user authentication, the system obtains call forwarding information for the user. The obtained call forwarding information is then evaluated for potentially fraudulent call forwarding configurations. For example, call forwarding to certain call forwarding numbers, or the use of different call forwarding types, may be indicative of fraud intended to undermine further user authentication.

Enablement of a voice channel being established between an IoT device and a controller through the use of a voice-line service system.

Embodiments described herein provide a dynamic voice over Internet Protocol (VoIP) audio quality management mechanism in real time, e.g., when a VoIP call is ongoing. Specifically, when a VoIP call has unsatisfactory audio quality, e.g., due to packet loss, jitter, etc., the dynamic VoIP audio quality management mechanism may redirect the VoIP traffic from the previous endpoint that initiates the VoIP session to a different endpoint within the same carrier. Upon the endpoint redirection, a new call leg is established, allowing re-negotiation or re-configuration of VoIP parameters. The re-negotiated or re-configured VoIP parameters may then be used to conduct the remainder of the VoIP call to improve the audio quality.

A mobile application and a method are described for servicing a second line service (SLS) based communication request originating from a subscriber's telecommunications device (TD) even if the call signal does not include sufficient information to identify the phone number from which the subscriber initiated the call. The method involves associating the SLS phone number of the subscriber, the primary number of the subscriber and the primary number of a third party via a special relationship number.