The IPG-40XG is an embedded Voice over IP (VoIP) PBX Server with Session Initiation Protocol (SIP) to provide IP extension phone connections for global virtual office of small-to-medium business (SMB) companys. Equipped with 4 x FXO ports, Ethernet LAN and WAN ports plus Life Line features, IPG-40XG integrates the telephony network and the data network into a manageable converged network to provide an efficient and economical PBX for global long distance voice communications.

IPG-40XG IP PBX works with various IP phones (Desktop, WiFi, Bluetooth, and DECT), VoIP gateways, and analog telephone adapters (ATA) to route calls among client phones, analog phones, and PSTN network. Call features such as conferencing, auto attendant, and voicemail can be seamlessly enabled for all phone devices. In addition, it also provides Internet access to all LAN devices through VPN NAT router.

IPG-40XG IP PBX provides call control and media relay services to SIP clients, and it performs many primary functions, such as SIP Registrar, SIP Outbound Proxy with media relay, SIP Gateways (FXO), SIP PBX for extension calls, Auto Attendant Interactive Voice Response (IVR), and Find-Me Conferencing.

IPG-40XG IP PBX has a built-in suite of PBX applications for supplemental services. This lowers down the total cost of a converged network enabled by IPG-40XG IP PBX than building separated infrastructures for legacy telephony network and data network. In addition, with a web-browsable interface to the data network configuration and voice service provisioning, IPG-40XG brings the manageability of both networks together to facilitate administration locally and/or remotely.

Note that IPG-40XG requires an IP address, a subnet mask, and its gateway Router IP address for its own use to connect to Internet. These three are available from your Internet service provider. IPG-40XG may enable PPPoE or DHCP features to automatically get an assigned dynamic IP from the ITSP. Please refer to Web Configurations for detailed information.

􀁺 SIP Server supports 50 user registrations and 25 concurrent calls

􀁺 Support 250 voicemail accounts

􀁺 Features Web-Calling

􀁺 SIP v1 (RFC2543), v2 (RFC3261) with MD5 authentication (RFC2069 and RFC 2617)

􀁺 Supports ITU-T G.711a, G.711u, GSM/MS-GSM, G.729A/B, VAD and CNG for Speech Codec

􀁺 Configurations by Web Browser

􀁺 Embedded NAT/DHCP Server

􀁺 PPPoE/DHCP Client for Dynamic IP plus NAT, VPN, DNS, and DDNS Clients

􀁺 Support STUN server and DMZ functions for NAT Traversal

􀁺 Support VPN function

􀁺 Provides Call Detail Record

􀁺 Support Call features; Call Forward/Waiting/Transfer/Hold, and Voice Conference Room

􀁺 Number Bonding and Call restrictions.

􀁺 Extension Pickup for Attendant

􀁺 Bill Rate Table with Voice Mail

􀁺 Interactive Voice Recording (IVR) Settings by XML

􀁺 Programmable Prompt messages

􀁺 On-Line Subscriber Status

􀁺 Remote Firmware Upgraded by HTTP Web Interface

􀁺 Auto Provision Settings

􀁺 Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info