Description
Grandstream HT801 Analog Telephone Adapter Dubai
Grandstream HT801 Analog Telephone Adapter Dubai is based on Grandstream’s leading SIP ATA/gateway technology with millions of units successfully deployed worldwide, therefore delivering HD voice quality in various scenarios and settings. Some of the advanced features packed into the powerful Grandstream HT801 Analog Telephone Adapter Dubai include automated provisioning for mass deployments and device management, impressive network performance, and powerful encryption with a unique security certificate per unit, among many others.
Grandstream HT801 Analog Telephone Adapter Features
1 FXS Port
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 HT801 Analog Telephone Adapter Dubai is equipped with 1 FXS port which allows 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 Grandstream HT801 Analog Telephone Adapter Dubai is equipped with 1 FXS port 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.
Fax Support
The Grandstream HT801 Analog Telephone Adapter Dubai 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 analogue signal used by fax machines must be converted into digital packets.
Cloud Management
Grandstream 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 into 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 HT801 Analog Telephone Adapter in Dubai uses this feature to eliminate delays, thus maximizing productivity.
Automated – Provisioning – TFTP/HTTP/HTTPS
The Grandstream HT801 Analog Telephone Adapter Dubai 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 HT801 Analog Telephone Adapter series in Dubai 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 HT801 Analog Telephone Adapter Dubai 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 HT801 ATA Dubai 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 HT801 Analog Telephone Adapter, 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 HT801 Analog Telephone Adapter Dubai.
Grandstream HT801 Key Features
- Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
- 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
- Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multi-language prompts, flexible dial plan and more
Grandstream HT801 Analog Telephone Adapter Specifications
Interfaces
- Telephone Interfaces: One (1) FXS port
- Network Interfaces: One (1) 10/100Mbps auto-sensing Ethernet port (RJ45)
- LED Indicators: POWER, INTERNET, PHONE
- Factory Reset Button: Yes
Voice, Fax, Modem
- 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: 5 REN: Up to 1km on 24 AWG
- Caller ID: Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID
- Disconnect Methods: Busy Tone, Polarity Reversal/Wink, Loop Current
Signalling
- Network Protocols: TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SSH, 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, SSH, TFTP, TR-069, secure and automated provisioning using AES encryption, Syslog
Security
- Media: SRTP
- Control: TLS/SIPS/HTTPS
- Management: Syslog support, SSH, remote management using the web browser