The Grandstream HT881 is an 8-port analog FXO gateway designed to integrate traditional PSTN lines with VoIP networks. It features 8 FXO ports and 1 FXS port, allowing businesses to connect analog phone lines and devices to IP-based systems.
🔧 Key Specifications
Telephony Interfaces
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FXO Ports: 8 RJ11 ports for connecting to PSTN lines
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FXS Port: 1 RJ11 port for analog devices (e.g., telephone or fax)
Network Interfaces
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Ethernet Ports: 2 × 10/100/1000 Mbps RJ45 ports
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Power Options: Supports Power over Ethernet (PoE) and 12V DC input
SIP & Voice Features
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SIP Profiles: Supports up to 3 SIP server profiles
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Voice Codecs: G.711 (A-law and μ-law), G.722, G.723.1, G.726-32, G.729A/B, iLBC, OPUS
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Fax Support: T.38 Fax-over-IP and G.711 pass-through
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Telephony Features: Caller ID display/block, call waiting, call transfer (blind/attended), call forwarding, call hold, do not disturb (DND), 3-way conferencing per port
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Echo Cancellation: Advanced line echo cancellation
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Jitter Buffer: Dynamic jitter buffer
Network & Security
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Network Protocols: TCP/IP, UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, TELNET, STUN, SIP (RFC3261), SIP over TCP/TLS, SRTP, TR-069
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QoS: Layer 2 (802.1Q VLAN, SIP/RTP 802.1p), Layer 3 (ToS, DiffServ, MPLS)
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Security: TLS and SRTP encryption for secure voice communication
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Provisioning: Supports TR-069 and XML configuration files for automated provisioning
Power & Environmental
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Power Supply: Input: 100-240V AC, 50-60Hz; Output: 12V DC, 1.5A
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Dimensions: Approximately 190 x 100 x 28 mm
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Weight: Approximately 0.46 kg
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Operating Temperature: 0°C to 50°C
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Humidity: 10% to 90% (non-condensing)
Compliance
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FCC, CE, RCM certifications
🔑 Key Features
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Supports 3 SIP profiles through 1 FXS port and 8 FXO ports
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Integrated high-performance NAT router
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Lifeline support: FXS port is hard-relayed to FXO port in case of power outage
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Automated & secure provisioning options using TR-069
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Supports T.38 Fax for reliable Fax-over-IP
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Strong AES encryption with security certificate per unit
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Failover SIP server automatically switches to secondary server if main server loses connection
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3-way voice conferencing per port
