Features:
2G GSM Gateway 850/900/1800/1900MHZ
Number of GSM channels (Max): 1
Network type: 850/900/1800/1900MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS, QoS, OpenVPN
NAT Traversal: Static NAT, STUN
Size: 110x70x24mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)
Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing
SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM, GSM to VoIP
GSM Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Incompatible with ATT network
NOTE: PLEASE CONTACT US FOR INSTANT MAP (MINIMUM ADVERTISED PRICING) DISCOUNTS ON THIS ITEM.
WARRANTY: 1-YEAR WARRANTY/ FINAL SALE
Features:
2G GSM Gateway 850/900/1800/1900MHZ
Number of GSM channels (Max): 1
Network type: 850/900/1800/1900MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS, QoS, OpenVPN
NAT Traversal: Static NAT, STUN
Size: 110x70x24mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)
Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing
SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM, GSM to VoIP
GSM Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Incompatible with ATT network
NOTE: PLEASE CONTACT US FOR INSTANT MAP (MINIMUM ADVERTISED PRICING) DISCOUNTS ON THIS ITEM.
WARRANTY: 1-YEAR WARRANTY/ FINAL SALE
