Mediant 600 - AudioCodes Mediant 1000 User Manual

Voice-over-ip (voip) sip media gateways
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Analog: The device's analog interface supports up to 24 analog ports (four ports per
analog module) in various Foreign Exchange Office (FXO) or Foreign Exchange
Station (FXS) configurations, supporting up to 24 simultaneous VoIP calls. The device
supports up to six analog modules, each module providing four analog RJ-11 ports.
The FXO module can be used to connect analog lines of an enterprise's PBX or the
PSTN to the IP network. The FXS module can be used to connect legacy telephones,
fax machines, and modems to the IP network. Optionally, the FXS module can be
connected to the external trunk lines of a PBX. When deployed with a combination of
FXO and FXS modules, the device can be used as a PBX for Small Office Home
Office (SOHO) users, and businesses not equipped with a PBX.
The device has enhanced hardware and software capabilities to ease its installation and to
maintain voice quality. If the measured voice quality falls beneath a pre-configured value,
or the path to the destination is disconnected, the device assures voice connectivity by
'falling' back to the PSTN. In the event of network problems or power failures, calls can be
routed back to the PSTN without requiring routing modifications in the PBX. Further
reliability is provided by dual Ethernet ports and an optional dual AC power supply.
The device supports various ISDN PRI protocols such as EuroISDN, North American NI2,
Lucent™ 4/5ESS, Nortel™ DMS100 and others. It also supports various ISDN BRI
protocols such as ETSI 5ESS and QSIG over BRI. In addition, it supports different variants
of CAS protocols for E1 and T1 spans, including MFC R2, E&M immediate start, E&M
delay dial / start, loop start and ground start.
The device provides a user-friendly, feature-rich embedded HTTP-based Web server for
remote configuration and management using a standard Web browser (such as
Microsoft™ Internet Explorer™), from anywhere in the world with IP connectivity to the
device.
1.2

Mediant 600

AudioCodes' Mediant 600 (hereafter referred to as device) is a cost-effective, wireline
Voice-over-IP (VoIP) Session Initiation Protocol (SIP)-based media gateway. It is designed
to interface between Time-Division Multiplexing (TDM) and IP networks in enterprises,
small and medium businesses (SMB), and CPE application service providers. Incorporating
AudioCodes' innovative VoIP technology, the device enables rapid time-to-market and
reliable cost-effective deployment of next-generation networks.
The device is based on VoIPerfect, AudioCodes underlying, best-of-breed, media gateway
core technology. The device provides superior voice technology for connecting legacy
telephone and PBX systems to IP networks, as well as seamlessly connecting IP-PBXs to
the PSTN. The device is fully interoperable with multiple vendors of IP-PBXs, IP Centrex
application servers, softswitches, gateways, proxy servers, IP phones, Session Border
Controllers and firewalls.
The device supports one or two E1/T1/J1 spans (including fractional E1/T1), or four to eight
ISDN Basic Rate Interface (BRI) ports. The device supports various ISDN PRI protocols
such as EuroISDN, North American NI2, Lucent™ 4/5ESS, Nortel™ DMS100 and others,
supporting different variants of CAS protocols, including MFC R2, E&M immediate start,
E&M delay dial / start, loop- and ground-start signaling. The device also supports various
ISDN BRI protocols such as ETSI 5ESS and QSIG over BRI. The device also provides
dual Ethernet 10/100Base-TX ports for IP redundancy.
Intelligently packaged in a stackable 1U chassis, the compact device can be mounted on a
desk or in a standard 19-inch rack.
The device provides a user-friendly, feature-rich Embedded HTTP-based Web server for
remote configuration and management using a standard Web browser (such as
Microsoft™ Internet Explorer™), from anywhere in the world with IP connectivity to the
device.
SIP User's Manual
20
Mediant 1000 & Mediant 600
Document #: LTRT-83303

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