-VoIP GSM/CDMA/UMTS gateway with 2 channels providing GSM/CDMA/UMTS network connectivity for soft switches and IP-PBXs.
 
Specifications:
Mobile: 2 GSM/CDMA/UMTS Channels
GSM Frequency: 850/900/1800/1900 MHz
UMTS Frequency: 850/1900 MHz, 850/2100 MHz, 900/2100 MHz
CDMA Frequency: 800 MHz
Antenna Splitter (4 in 1):
Protocol: SIP, 1AX2
Transport: UDP, TCP, TLS, SRTP
Video Codec: G.711 (alaw/ulaw), G.722, G.726, G.729A, GSM, ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
DTMF Mode: RFC2833, SIF INFO, In-Band
Console Port:
Calling Type: Termination (VoIP to GSM/CDMA/UMTS), Origination (GSM/CDMA/UMTS to VoIP)
LAN: 1 10/100Mbps
Network Protocol: FTP, TFTP, HTTP, SSH
NAT Traversal: Static NAT, STUN
Network: DHCP, DDNS, Firewall, OpenVPN, Static IP, QoS, Static Route, VLAN
Dimension (L x W x H) (mm): 213 x 160 x 44
Power Supply: AC 100-240V
Operating Range: °C to 40°C, 32°F to 104°F
Storage Range: -20°C to 65°C, -4°F to 149°F
Humidity: 10-90% non-condensing
 
Features:
1 Stage/2 Stage Dial
Call Back
Call Duration Limitation
Call Status Display
Call Waiting
Carrier Selection: Auto/Manual
Firmware upgrade by HTTP/TFTP
GSM/CDMA/UMTS Ports Group
Manage
Incoming / Outgoing Routing Rules
Network Attack Alert
Open API for SMS and USSD
PIN Modify
Send Bulk SMS
SIP Per Mode: Support
SIP server for IP phones: Support
SMS Center
System Logs
VoIP Trunk Group
White List and Black List
Balance Alarm
Call Detail Record (CDR)
Call Progress Tone Generation
Call Transfer
Call ID/CLIR
Configure back/restore
Gain Adjustment
Hotline
IP Blacklist
NTP
Packet Capture
Real Open API Protocol (Based on Asterisk)
Session Timer
SIP Response Code Switch
SIP Trunk: Support
SMS Sending and Recieving
USSD
Web base configuration