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1. Technical parameters
- Dimensions
- Weight
- Operating position
- Operating condition
- Power supply
VoIP
Ethernet – 10/100Mb with standard BaseT and 100BaseTx, connector RJ45
•
SIP connection: P2P or IPPBX SIP server – tested with Cisco Call Manager, Alcatel OMNI PCX, Asterisk,
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Nexspan, Panasonic...
2 VoIP channels (2 IP addresses)
•
Codec audio : G711u, G711a, G726, GSM
•
VAD (Echo cancellation)
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Protocols: IP, TCP, UDP, http, TELNET, SIP, RTP
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Web server for remote management
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WEB – firmware upgrade
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Remote connection with DTMF dial in
•
nd
connection to 2
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Caller Id number (CLIP)
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incoming call restriction from GSM network
•
outgoing call restriction to GSM network
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Priority connection through either the 1
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Smart Call back – automatic incoming calls routing up CLIP
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Direct access – assign of IP address to GSM channels
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Echo canceller – switching ON/OFF
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OGM* Option module for recording DISA voice message*
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PIN SIM card protection
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GSM
GSM 900 (class 4 – 2 W)
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GSM 1800 (class 1 – 1 W)
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Antenna connector SME/SMA, 50
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SIM card: 3/1,8 V
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2 GSM channels
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2. Setting operational Mode (P2P or SIP Server)
The Link Gate SIP can be configured in 2 different modes: P2P or SIP Server.
In P2P mode, the Link Gate SIP behaves as a sample GSM gateway. In case of incoming call on the GSM
•
module, the Link Gate SIP will convert it in SIP and transfer it to the SIP server (IPBX). For an out going call,
the Link Gate SIP process a GSM call when SIP call is receives over IP addresses.
In SIP Server mode, the Link Gate SIP acts as a SIP server and allow recording up to 10 IP/SIP phones. It
•
is also possible to register the Link Gate SIP on an existing IPBX.
P2P Mode
Linkcom France
11 rue du Soleil Levant 92140 CLAMART
133 x 233 x 60 mm
850 g
various
temperature: +5°C ÷ +40° C
Humidity: 10% ÷ 80% p i 30° C
9-15V ss or 8-12Vst, 1.5 A
operators after time out (adjustable) for direct dialling in
st
or the 2
nd
GSM module (LCR)
Copyright © 2008 – Linkcom – Tous droit réservés
SIP Server Mode
3
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