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.
  

 

l         SIP Server supports 50 user registrations and 25 concurrent calls l         Support 250 voicemail accounts l         Features Web-Calling l         SIP v1 (RFC2543), v2 (RFC3261) with MD5 authentication (RFC2069 and RFC2617) l         Supports ITU-T G.711a, G.711u, GSM/MS-GSM, G.729A/B, VAD and CNG for Speech Codec l         Configurations by Web Browser l         Embedded NAT/DHCP Server l         PPPoE/DHCP Client for Dynamic IP plus NAT, VPN, DNS, and DDNS Clients l         Support STUN server and DMZ functions for NAT Traversal l         Support VPN function l         Provides Call Detail Record l         Support Call features; Call Forward/Waiting/Transfer/Hold, and Voice Conference Room l         Number Bonding and Call restrictions. l         Extension Pickup for Attendant l         Bill Rate Table with Voice Mail l         Interactive Voice Recording (IVR) Settings by XML l         Programmable Prompt messages l         On-Line Subscriber Status l         Remote Firmware Upgraded by HTTP Web Interface l         Auto Provision Settings l         Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info PBX Features l       Support call hold, IP Call broadcasting with feature phones l       Built-in in-line call transfer l       Unconditional, unavailable, busy, call forward l       Per-calling-number forward and rejection l       Group-based call pick-up l       Multi-room meet-me conference l       Auto-attendant l       Voice mail system l       Call privilege grouping l       FXO interface for PSTN l       Inbound/Outbound l       FXO disconnection tone detection l       FXO hunt group l       Caller ID detection l       Echo cancellation l       Support 5 SIP trunk l       In-band/RFC2833/SIP-INFO DTMF translation l       Intra-PBX stackable trucking over Ethernet l       FWD/Vantage account sharing for extensions l       Interoperable with Cisco call manager, CCME; IOS SIP gateway; Unity CUE,79XX, ATA l       Call admission control for wire/wireless phones l       Music on hold l       Direct line

	IPG-402G : 4 Port SIP VoIP Gateway Embedded IP PBX ( 2FXO+2FXS / 4 FXO ) IPG-404G : 4 Port SIP VoIP Gateway Embedded IP PBX (4FXO)

 

Meet-me Conference

l       Multiple rooms with configurable number and PIN

l       Hot key to leave conference

Trunk

l       Auto trunk selection

l       Size specific trunk

l       Call Barring/Transit Call

l       PSTN to remote site

l       Direct outward dial

Internal Line

l       Call pickup group/ Group hunting/ Hot line/ Group Ringing

Extension

l       Call Transfer/ Call forward/ forward Me/ Call hold/ Call Park/ Do not disturb/
Call pickup/ Conference call

System

l       Music on hold

l       Call detail recording

l       Built-in dialer

l       External voice mail

l       Networking &stackable

l       Support private IP network
 

Voice Mail

l       User PIN

l       Multilingual

l       Multi-folder archive

l       Fast-forward/Rewind/ Undelete

l       Personal reception on unavailability/ busy

l       Voicemail forward

l       Reply call or new call in voicemail menu
 

Special Support

l       Built-in WEB Call

l       Support CBCOM

Support Standards

      RFC 3261, RFC 3311, RFC 3515, RFC 3265, RFC 3892, RFC 3361, RFC 3842,
      RFC3389, RFC 3489, RFC 3428, RFC 2327, RFC 2833, RFC 2976, RFC 3263,
      RFC3264

Network Management

l       DHCP/PPPoE/Static IP

l       LAN IP and net mask specification

l       Firewall on predefined services

l       DNS and dynamic DNS
 

NAT

l       Auto NAT discovery and traversal

l       RTP proxy

l       RTP port range designation

SIP Proxy

l       Proxy server

l       Call-based MD5 authentication

l       NAT traversal for clients

l       Outbound proxy with or without WAN

l       Inter-proxy call hand-off

SIP Registration

l       Static/ Dynamic registration

l       MD5 authentication

l       Authentication proxy to external registrars

l       Configuration PBX Caller ID

l       User profile

l       Handle loose RFC-compliant phones

l       Resilient message retry  mechanism

l       Seeding historical registrations

Relational Provision

l       Logical partition/relation on user and trunks

l       Logical provision on outgoing and incoming calling search scopes

l       Rich dial-plan expressiveness through route patterns

l       Object-oriented provisioning paradigm administration

l       Web-based configuration

l       CDR

l       Extension status display

l       Network time protocol time synchronization

l       Real time clock setting

l       DHCP server

l       Configuration time zone

l       Firmware upgrade through web interface

Auto Attendant

l       Configuration greeting

l       Key to reach operator

l       Timeout interval and timeout action

l       Music on ringing extensions

l       Forward to voice mail on no-answer