Description
Gandstream HT814 Analog Telephone Adapter Oman
The Gandstream HT814 Oman powerful gateway is integrated with state-of-the-art encryption with a unique security certificate per unit, automated provisioning which allows for easy mass deployments, device management, as well as and impressive network performance. Grandstream HT814 Oman is designed using Grandstream’s most popular SIP ATA/gateway technology, with millions of units successfully deployed worldwide. It is a powerful ATA that features exceptional voice quality which is highly applicable in various applications and environments.
Features
4 FXS Ports
FXS stands for Foreign Exchange Subscriber or Foreign Exchange Station. This port in gateways is used to connect an analog endpoint like a fax machine or an analog phone. A device that has an FXS port is capable of allowing an analog device to connect to a VoIP network. The Grandstream HT814 Oman is equipped with 4 FXS ports which allow for easy connection of your analog device to a VoIP network so that you can enjoy making and receiving fast calls over the internet.
3-Way Voice Conferencing per port
The Gandstream HT814 Oman is equipped with 8 FXS ports, each of which supports 3-way HD audio conferencing. This enables you to seamlessly connect and collaborate with colleagues from different geographical locations. It is a cost-effective means of communication that maximizes efficiency as well as boosts productivity.
Integrated NAT Router
Gandstream HT814 Analog Telephone Adapter Oman comes with an integrated NAT router with dual Gigabit Ethernet ports which allow you to share a broadband connection between multiple devices on your LAN without interfering with connection speed.
Fax Support
The Grandstream HT814 Oman is configured to support T.38 Fax which enables you to create Fax-over-IP. 38 is an ITU protocol that dictates how to fax audio traverses a packet-switched network, such as the Internet or a company’s LAN/WAN, reliably. In order for fax audio to traverse a packet-switched network like the Internet, the analog signal used by fax machines must be converted into digital packets.
Cloud Management
Gandstream HT814 Oman provides a reliable and secure cloud service to deliver connectivity to its users at cheaper maintenance costs while enhancing operation and maintenance efficiency. The cloud service on the internet enables customers to gain direct access to Grandstream devices available on private networks such as IP-PBXs, audio gateways, and IP phones, therefore allowing simple remote maintenance and management of Grandstream devices. This kind of cloud service is specifically designed to meet the needs of large-scale installation, configuration, and maintenance and operation. Auto-provisioning and configuration backup, online update, real-time monitoring and alert enable users to achieve efficient operation and maintenance, thus maximizing productivity in organizations.
Layer 2 & Layer 3 QoS Control
Quality of Service (QoS) is a strong feature that is integrated in routers and switches so as to control traffic flow thus delivering an overall smooth performance improvement for critical network traffic. It simply prioritizes more important traffic by letting it pass first. Whenever there are high volumes of traffic, the Grandstream HT814 VoIP Gateway in Oman uses this feature to eliminate delays, thus maximizing productivity.
Automated – Provisioning – TFTP/HTTP/HTTPS
The Grandstream HT814 VoIP Gateway Oman has the ability to deploy an information technology or telecommunications service by using embedded pre-defined procedures that are carried out electronically without requiring any human intervention, thus making it very reliable and convenient.
HD Voice Quality
The Grandstream HT814 Oman supports a number of voice compression codecs including 711 with Annex I (PLC) and Annex II (VAD/CNG), G.723.1, G.729A/B, G.726, iLBC,and OPUS. It also uses noise suppression technologies such as Echo Cancellation and Jitter Buffer to deliver crystal clear sounds while filtering out any background noises which may interfere with communication during voice calls.
Maximum Security
Grandstream HT814 Oman uses a number of leading security protocols including TLS and SRTP security encryption technology to safeguard your VoIP accounts and calls from access by unauthorized persons and devices. This prevents the loss of important data or corruption of files due to access by malicious hackers.
Call Features
- Call Transfer : This advanced feature of the Grandstream HT814 gateway allows the user to relocate an inbound call to another phone or messaging system by using a dedicated call transfer button, or software that has been configured for use on the Gateway.
- Call Waiting : You can hear another incoming call when you are already on an active phone call (beep). With the Grandstream HT814 VoIP Gateway, you can also turn off call waiting so that incoming calls are directly sent to voicemail during moments you are active on another phone call.
- Call Holding : You can easily place an active phone call on hold in order to make or pick another incoming call using the premium Grandstream HT814 VoIP Gateway Oman.
Key Features
- Supports 2 SIP profiles through 4 FXS ports and dual Gigabit ports
- Includes a built-in NAT router that can handle routing speeds up to 100MBps
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning options include TR-069 and XML config files
- Supports 3-way voice conferencing
- Failover SIP server automatically switches to secondary server if main server loses connection
- Supports T.38 Fax for creating Fax-over-IP
- Supports a wide range of caller ID formats
- Use Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
Specifications
Telephone Interfaces | Four (4) FXS ports |
Network Interfaces | Two (2) 10/100/1000Mbps RJ45 ports |
LED Indicators | POWER, LAN, WAN, PHONE1, PHONE2, PHONE3, PHONE4 |
Factory Reset Button | Yes |
Telephony Features | Caller ID display or block, call waiting, flash, blind or attended transfer, forward, hold, do not disturb, 3-way conference |
Voice Codecs | G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.723.1, G.729A/B, G.726, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation |
Fax Over IP | T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through |
Short/Long Haul Ring Load | 2 REN, up to 1km on 24AWG line |
Caller ID | Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID |
Disconnect Methods | Busy Tone, Polarity Reversal/Wink, Loop Current |
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 |
QoS | Layer 2 (802.1Q VLAN, SIP/RTP 802.1p) and Layer 3 (ToS, Diffserv, MPLS) |
DTMF Method | In-audio, RFC2833 and/or SIP INFO |
Provisioning and Control | HTTP, HTTPS, TELNET, TFTP, TR-069 , secure and automated provisioning using AES encryption, syslog |
Media | SRTP |
Control | TLS/SIPS/HTTPS |
Management | Syslog support, telnet, remote management using web broswer |
Universal Power Supply | Input : 100-240VAC, 50-60Hz Output : 12V/1A |
Environmental | Operational :
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Dimension and Weight | 28.5 x 130 x 90 mm (H x W x D) Weight : 353.33g |
Compliance | FCC/CE/RCM |