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Suitable for small to
medium enterprises and easy to install, configure, and maintain the
Dialogic 1000 is compatible with a variety of popular PBX manufacturers
including Alcatel, Avaya, Ericsson, Fujitsu, Mitel, NEC, Nortel, and Siemens.
Applications
-Centralized VoIP and FoIP application servers, including IP-based voice mail and unified messaging -IVR and announcements -IP PBX -VoIP extension to branch offices -Contact centers
Functional Description
The DMG1000 Gateways each contain eight digital PBX emulation interfaces and a 10/100 BaseT Ethernet connection for connecting to a LAN. An analog loop start unit designed for voice mail and unified messaging applications is also available to connect to PBXs that do not have an appropriate digital interface. The analog loop start unit supports integration via in-band signaling (DTMF or FSK) or serial protocols (SMDI, MCI, and MD-110). The DMG1000 Gateways provide a simple, costeffective transition to voice and data convergence for enterprises with PBXs. Connected externally, they offer an IP solution that works with current legacy equipment. They support SIP-based applications as well as T.38 for fax transmissions over IP (FoIP).
Gateway unit features include:
Voice over Internet Protocol (VoIP) -
Supports SIP per RFC 3261. Uses Real-time Transport Protocol/Real-Time
Control Protocol (RTP/RTCP) for delivery of voice over the LAN or WAN
IP security - Supports TLS for SIP messages, SRTP for media stream, and HTTPS for web interface
Enhanced voice processing - Supports a variety of compression algorithms, including G.711 A-law and μ-law, G.723.1, and G.729AB
T.38
Fax over Internet Protocol (FoIP) - Emulation units transcode fax from
T.30 fax protocol, supporting V.17, V.21, V.27, and V.29 modulation
schemes, to T.38 for transmission over a packet network
Hot swap - Allows gateway units to be added or removed without affecting other gateway units
Web
server interface - Each gateway unit is delivered with a web server
interface, allowing configuration and software upgrades via a web
browser
The DMG1000 Gateways can be used to connect IP telephones to a legacy
PBX, integrate network-hosted applications with the PBX, extend the PBX
to branch offices, and integrate various voice and call processing
capabilities in an enterprise LAN or WAN environment. Using exclusive
PBX network interfaces (emulating), these media gateway appliances
provide exceptional IP to PBX integration capabilities to protect an
investment in legacy telecom equipment.
Call Routing
The PBX-IP
media gateway routes calls from the switched network to a VoIP
destination on the IP network. Conversely, it routes calls from the IP
network through a switch port to a destination telephone number on the
switched network. The gateway supports the following call routing
options:
- User-configurable list of VoIP servers - IP load balancing - IP fault tolerance


Interfaces
DC power Serial port for diagnostics or serial protocol support 4 telephony ports Ethernet port Reset switch
Status Indicators
10/100BaseT Full/half duplex RX/TX traffic Ethernet link state Ethernet collision
Technical Specifications
PBX Interface Number of ports: 4 and 8 port analog units, and 8 port Digital PBX emulation units - Use multiple gateway units for higher port counts
Connectors: 8 shielded female RJ-45 jacks
Network Interface: 10/100 Base-T Ethernet LAN port
Connector: 1 shielded female RJ-45 jack for LAN
VoIP Protocols: SIP per RFC 3261, RTP/RTCP for delivery of voice
FoIP Protocol: T.38 FoIP
Emulation units transcode fax from T.30 fax protocol, supporting V.17,
V.21, V.27, and V.29 modulation schemes, to T.38 for transmission over
a packet network
Voice Support: G.711 μ-Law and A-Law, G.723.1, G.729AB. Silence suppression with comfort noise, G.168 automatic echo cancellation
Quality of Service: Type of Service (ToS), IP precedence
Configuration and Management: SNMP v1 Read-only for alarm reporting, Web GUI With context-sensitive Help facility, Telnet, BOOTP client and TFTP client
Call Routing: User
configuration list of VoIP endpoints, IP load balancing, IP fault
tolerance, Supports configuration of a backup SIP proxy server
IP Security: TLS for SIP messages, SRTP for media stream, HTTPS for web interface
Power Requirements: Line voltage 90 VAC to 264 VAC, Frequency 47 Hz to 63 Hz
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