1/2/4/8 ports FXS gateway sip ATA gateway voip ip phone
USD $84 - $84 /Piece
Min.Order:1 Piece
Skyline (Shenzhen) Technology Co., Ltd.
1/2/4/8 ports FXS gateway sip ATA gateway mediant 2000 voip gateway
1/2/4/8 ports FXS gateway sip ATA gateway mediant 1000 voip gateway
1/2/4/8 ports FXS gateway sip ATA gateway audiocode mediant 1000
1/2/4/8 ports FXS gateway sip ATA gateway audiocodes mediant
1/2/4/8 ports FXS gateway sip ATA gateway audiocode mediant 2000
1/2/4/8 ports FXS gateway sip ATA gateway audiocodes gateways
4 ports fxs gateway voip ip phone VoIP FXS Gateway HT-842R is a telephone extension to the IP network.It offers a traditional telephone line (PSTN) interface to an analog telephone,PBX line extension, or a fax machine.Its WAN port interface allows access to the IP network in order to offer voice and fax services.It is a great way for turning a traditional PBX to access the low cost VoIP services and for deploying VoIP service by an ISP.An additional Ethernet port allows broadband connection for the existing PC or other network device without buying additional network equipment. It is an ideal low cost product for SME and SOHO IP telephony application. | |
Product overview | 4 FXS ports VOIP Gateway |
Open Standard VoIP Protocols (SIP&H.323) | |
Provide 1 RJ-11 port for a POTS phone or a PBX's trunk line | |
Provide 2 RJ-45 Ethernet Ports | |
Support call forward/transfer/hold, phone book | |
Support G.711 A/μ law, G.729A/B, G.723.1 Codecs | |
Support QoS, NAT transversal and router function | |
Support VAD, CNG, EC | |
Highlights | Built-in H.323 V4 and SIP V2 Protocols. |
Low cost | |
Hight performance router | |
Caller ID | |
Auto provisioning and firmware updates | |
Encryption transversal | |
Multiple languages | |
Key Features | Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2) |
Two 10/100 Ethernet for WAN / LAN connections | |
Peer-to-Peer IP Calls | |
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer | |
Line Echo Cancellation | |
VLAN and QoS support | |
NAT Transversal and Router functions | |
Voice prompts, HTTP Web, Auto Provision support for configuration and updates | |
Highly stable embedded Linux operating system in high performance ARM 9 Processor | |
Basic Feature | 1 RJ-11 FXS port for traditional phone set or PBX's trunk line |
LEDs for Power, Ready, Status, WAN, PC, FXS | |
Call Forward, Call Hold, Call Transfer Dial Plan | |
Caller ID | |
Enhanced Features | Dynamic selection of codec |
Advanced jitter buffer | |
Automatic traversal of NAT and firewall | |
VLAN / Qos | |
Router | |
Echo cancellation for Speakerphone | |
Comfort noise generation (CNG) | |
Voice activity detection (VAD) | |
Auto provisioning (requires auto provisioning server) | |
On line firmware upgrade | |
Multi-language support: English and Chinese | |
Supported Standards | ITU: H.323 V4, H.225, H.235, H.245, H.450 |
RFC 1889 - RTP/RTCP | |
RFC 2327 - SDP | |
RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals | |
RFC 2976 - SIP INFO Method | |
RFC 3261 - SIP | |
RFC 3264 - Offer/Answer model with SDP | |
RFC 3515 - SIP REFER Method | |
RFC 3842 - A Message Summary and Message Waiting Indicator | |
RFC 3489 - Simple Traversal of User Datagram Protocol (UDP) | |
Through Network Address Translators (NATs) | |
RFC 3891 - SIP "Replaces" Header | |
RFC 3892 - SIP Referred-By Mechanism | |
draft-ietf-sipping-cc-transfer-04 - Session Initiation Protocol Call Control - Transfer | |
Codec: G.711 (A/μ law), G.729A/B, G.723.1 | |
DTMF: RFC 2833, In-band DTMF, SIP INFO | |
Free Software | Auto Config Server |
Relay Server | |
Remote Access | |
Billing | |
Hardware Specifications | Operating temperature: 10°C to 40°C (50°F to 104°F) |
Storage temperature: 0°C to 50°C (32°F to 122°F) | |
Size: 167mm (W) x 267mm (L) x 65mm (H) | |
Weight:0.56kg (Including AC/DC Adapter) | |
Power: 24V 500mA | |
Warranty: 1 year |
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