Grandstream HT818 V2
The Grandstream HT818 V2 is an 8-port analog telephone adapter and VoIP gateway designed to connect multiple analog phones, fax machines, and compatible legacy telephony devices to modern SIP-based communications systems. Eight RJ11 FXS ports, two Gigabit Ethernet interfaces, an integrated NAT router, wideband HD voice codecs, and advanced telephony features make it a practical choice for businesses, service providers, hospitality environments, and multi-site deployments transitioning analog endpoints to VoIP.
Built for both individual installations and large-scale managed deployments, the HT818 V2 supports two SIP profiles, T.38 Fax-over-IP, 3-way conferencing, automatic SIP server failover, secure automated provisioning, and Grandstream UCM Zero Configuration. Flexible network-port operation, strong TLS and SRTP security, and centralized device-management capabilities help simplify installation and ongoing administration across professional VoIP environments.
Key Features
- 8 FXS Ports: Connects up to eight compatible analog telephones, fax machines, or other analog endpoints to a SIP-based VoIP system.
- 2 SIP Profiles: Supports two SIP profiles for flexible account and service-provider configuration across the eight FXS interfaces.
- Dual Gigabit Ethernet: Two 10/100/1000 Mbps RJ45 ports support high-speed network connectivity and flexible WAN/LAN operation.
- Integrated NAT Router: Built-in routing capabilities allow the HT818 V2 to provide network connectivity alongside analog-to-VoIP conversion.
- Wideband HD Voice: Supports G.722, Opus, and additional industry-standard codecs for high-quality voice communications.
- T.38 Fax Support: T.38 Group 3 Fax Relay and automatic G.711 fax pass-through support compatible Fax-over-IP deployments.
- 3-Way Conferencing: Supports 3-way voice conferencing on each FXS port.
- Advanced Calling Features: Supports call transfer, forwarding, waiting, hold, do not disturb, caller ID, message waiting indication, and flexible dial plans.
- SIP Server Failover: Automatically switches to a configured secondary SIP server if the primary server becomes unavailable.
- Secure Provisioning: Supports automated provisioning through TR-069, XML configuration files, HTTPS, TFTP, FTP, and FTPS.
- Grandstream UCM Integration: Supports Zero Configuration provisioning with compatible Grandstream UCM Series IP PBXs.
- Centralized Management: Supports Grandstream device-management tools for scalable provisioning, monitoring, and administration.
Applications & Use Cases
- Analog-to-VoIP Migration: Allows organizations to continue using compatible analog telephones while moving calling services to SIP-based VoIP infrastructure.
- Business Offices: Connects multiple analog desk phones to hosted VoIP services or compatible IP PBX systems.
- Hospitality: Supports hotels and similar properties that need to integrate multiple analog room phones with modern IP communications.
- Service Provider Deployments: Automated provisioning and centralized management capabilities support high-volume ATA installations.
- Grandstream UCM Systems: Works with compatible Grandstream UCM Series IP PBXs and supports Zero Configuration provisioning.
- Fax-over-IP: T.38 and G.711 fax support allow compatible analog fax machines to operate in appropriately configured VoIP environments.
- Multi-Tenant & Multi-Site Environments: Eight FXS interfaces make the gateway useful for apartments, branch offices, healthcare facilities, schools, and other locations with multiple analog endpoints.
- Managed VoIP Deployments: Remote configuration and provisioning tools help MSPs, IT teams, and service providers manage distributed installations.
Technical Specifications
Interfaces & Connectivity
- Telephone Interfaces: 8 × RJ11 FXS ports
- SIP Profiles: 2
- Network Interfaces: 2 × RJ45 Ethernet ports
- Ethernet Speed: 10/100/1000 Mbps
- Integrated Router: NAT router
- NET1 Default Mode: WAN / DHCP client
- NET2 Default Mode: LAN / DHCP server
- Configurable Network Ports: NET1 and NET2 operating roles can be changed through the web interface
- Factory Reset: Supported
- LED Indicators: Power, NET1, NET2, Phone 1 through Phone 8
Voice & Telephony
- Voice Codecs: G.711, G.722, G.723.1, G.729A/B, G.726-32, iLBC, Opus
- Packet Loss Concealment: G.711 Annex I
- Voice Activity Detection: G.711 Annex II VAD/CNG
- Jitter Buffer: Dynamic
- Echo Cancellation: Advanced line echo cancellation
- 3-Way Conferencing: Supported
- Call Transfer: Blind and attended
- Call Forwarding: Supported
- Call Waiting: Supported
- Call Hold: Supported
- Do Not Disturb: Supported
- Caller ID: Bellcore Type 1 and 2, ETSI, BT, NTT, and DTMF-based CID
- Dial Methods: DTMF and pulse
- DTMF Methods: In-audio, RFC2833, SIP INFO
- Disconnect Methods: Busy tone, polarity reversal/wink, loop current
Fax & Analog Line
- Fax over IP: T.38 compliant Group 3 Fax Relay
- Maximum T.38 Fax Rate: Up to 14.4 kbps
- Fax Pass-Through: Automatic G.711 fax pass-through
- Ring Load: 2 REN
- Maximum Line Distance: Up to 1 km using 24 AWG line
- GR-909 Line Testing: Supported
Network & SIP
- SIP: SIP v2.0 / RFC3261
- SIP Transport: TCP and TLS
- Network Protocols: TCP/IP, UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SSH, Telnet, STUN, SIP, SNMP, TR-069, IMS/3GPP, IPoE
- Layer 2 QoS: 802.1Q VLAN and SIP/RTP 802.1p
- Layer 3 QoS: ToS, DiffServ, MPLS
- Traffic Shaping: Supported
- SIP Server Failover: Automatic failover to configured secondary SIP server
Provisioning, Management & Security
- Provisioning: HTTP, HTTPS, FTP, FTPS, SSH, TFTP, and TR-069
- Automated Provisioning: Supported
- Configuration Files: XML configuration support
- Management: Web interface, SSH, syslog, and compatible centralized Grandstream management tools
- Media Security: SRTP
- Control Security: TLS, SIPS, HTTPS
- AES Encryption: Supported
- Device Certificate: Unique security certificate per unit
- UCM Zero Configuration: Supported with compatible Grandstream UCM Series IP PBXs
Power & Physical
- Power Adapter Input: 100-240VAC, 50/60Hz
- Power Adapter Output: 12VDC / 1.5A
- Dimensions: 36 × 120 × 180 mm
- Weight: 356 g
- Operating Temperature: 0°C to 40°C
- Storage Temperature: -10°C to 60°C
- Humidity: 10% to 90% non-condensing
- Compliance: FCC, CE, RCM
Package Includes
- Analog Telephone Adapter: 1 × Grandstream HT818 V2
- Power Adapter: 1 × 12V power adapter
- Ethernet Cable: 1 × Ethernet cable
- Documentation: 1 × Quick Installation Guide
Compatibility & Buying Information
- FXS Gateway: The HT818 V2 provides eight FXS ports for connecting compatible analog telephones, fax machines, and other analog endpoint devices to SIP-based communications systems.
- Not an FXO Gateway: The FXS ports are designed for analog endpoint devices and are not intended for connecting incoming analog PSTN telephone lines.
- SIP Service Required: Calling requires a properly configured SIP account, hosted VoIP provider, or compatible SIP-based IP PBX.
- Network Ports: Both NET1 and NET2 support Gigabit Ethernet. NET1 defaults to WAN/DHCP client operation and NET2 defaults to LAN/DHCP server operation, with configurable port roles.
- Integrated Routing: The built-in NAT router can provide network routing in addition to analog-to-VoIP conversion when required by the deployment.
- Grandstream UCM Compatibility: Supports deployment with compatible Grandstream UCM Series IP PBXs and Zero Configuration provisioning.
- Fax Compatibility: Supports T.38 Group 3 Fax Relay and G.711 fax pass-through. Actual fax performance depends on the fax device, SIP provider, PBX configuration, and network conditions.
- Power: Uses the included 12VDC / 1.5A power adapter. Grandstream instructs users to use the qualified adapter supplied with the device.
- V2 Platform: Use firmware, documentation, and configuration resources specifically intended for the HT818 V2 / HT81x V2 hardware platform rather than resources for the original HT818 hardware.
Manufacturer
GRANDSTREAM
Manufacturer Part
HT818 V2
Warranty
12 MONTH
UPC
6947273705543
Condition
NEW