SC-895N GSM VoIP Terminal with 8SIM,
1WAN for 8 channels use, SMS, Quad band
(Support NAT, Auto Hunting)

SC-895N GSM VoIP Terminal has 8 channels for call termination (VoIP to GSM) and
origination (GSM to VoIP) simultaneously. It is SIP based, compatible with Asterisk Soft
Switch IP PBX, SIP Proxy Server, VoipBuster, 3CX etc. Which enable to make 8 calls from
IP phones to GSM networks and from GSM networks to IP Phone at the same time. SC-895N
support Built in Dial peer Server, Stun Server & NAT Function
It is like a traditional PBX. 5060 is a delegate NUMBER
Which has 8 lines: 5064, 5066, 5068, 5070, 5072, 5074, 5076, 5078
The call automatically switches from a busy line to available line.
So user just send call to 5060 port from Asterisk/ IP PBX
*** 5060 Port can be changed to any port as user’s need***
Major Function:
1. VoIP (SIP) – GSM conversion
2.
50 sets of LAN --> MOBILE routes setting; 50 sets of MOBILE --> LAN routes setting.
- Support one stage dialing:
*When LAN phone and SC-895 both register SIP proxy Server or Asterisk or VoipBuster,
Users can dial any destination number from LAN phone directly.
*** Please note, SIP proxy Server, Asterisk need to have the route of destination number.
VoipBuster must have credit in deposit ***
- Support free mode - 2 stages dialing & assigned mode - 1 stage dialing
3. Voice response for setting and status (dial in from mobile).
4. For call termination (VoIP to GSM) and origination (GSM to VoIP).
5. Standard SIP (RFC2543,RFC3261) protocol, Communicates with other gateway or PC
6. Receive SMS and Send SMS (CDMA VoIP Terminal: SMS feature is not available)
7. Allows your program to Send/receive SMS with AT Command
8.. Support Call Back feature
9. All functions can be set on web.
Specification:
Protocols: SIP (RFC2543, RFC3261)
TCP/IP: IP/TCP/UDP/RTP/RTCP/, CMP/ARP/RARP/SNTP, DHCP/DNS Client,
IEEE802.1P/Q, ToS/DiffServ, NAT Traversal, STUN, uPnP, IP Assignment, Static IP,
DHCP, PPPoE
Codec: G.711 u-Law, G.711 a-Law, G.729

Major Function:
VoIP (SIP)- GSM conversion
50 sets of LAN -> MOBILE routes setting;
50 sets of MOBILE -> LAN routes setting
Voice response for setting & status (dial in from mobile)
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP (RFC2543, RFC3261) protocol,communicates with other gateway or PC
To be set on web
Specification:
SIP protocol: (RFC2543, RFC3261)
TCP/IP: IP/TCP/UDP/RTP/RTCP/、CMP/ARP/RARP/SNTP DHCP/DNS Client、IEEE802.1P/Q、ToS/DiffServ、NAT Traversal、STUN、uPnP、IP Assignment、Static IP、DHCP、PPPoE
Codec: G.711 u-Law、G.711 a-Law、G.723.1 (5.3k) G.723.1 (6.3k)、G.729A、G.729A/B、Voice Quality VAD、CNG、AEC、LEC、Packet loss
GSM Frequency:
Quad band 900/1800/1900/850MHz