WellTech M1 - GSM Gateway 1 Port

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Rp 2.460.000,00

Sinopsis

WELLTECH M1 GSM Gateway 1 Port


WellGate M1 is single SIM GoIP GSM SIP gateway which connects SIP VoIP with GSM BTS (base station). This device is suitable for office IP-PBX application or office to branch office to call between PSTN Line office and IP Call. Or, connect to IP Telephony SIP Service provider from/to GSM mobile network.

WellTech M1 - GSM Gateway 1 Port

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Product Overview

WellTech M1 is single SIM GoIP GSM SIP gateway which connects SIP VoIP with GSM BTS (base station). This device is suitable for office IP-PBX application or office to branch office to call between PSTN Line office and IP Call. Or, connect to IP Telephony SIP Service provider from/to GSM mobile network.

The traditional PSTN land line connects to FXO gateway causes many installation issues, for instance, disconnect one detection, loop current drop detection, Caller ID analog signal detection (DTMF, FSK or Bellcore type caller ID). Moreover, FXO gateway is not able to detect remote user answer call exact time while FXO gateway makes a call out. As a result, the billing time is not accurate. Some area is not able to provide land line and only mobile signal provided. WellGate M1 is single GSM SIP gateway which solves above issues.

 

WellTech M1 - Technical Specifications
 

Interface

  • 1 GSM SIM Card Tray
  • Ethernet Port (RJ-45, 10/100 base-T)
  • 1 LAN interface, Connect to local network
  • 1 WAN interface, Connect to IP network
  • DC +12V power input Jack
  • Reset key to return Factory setting
  • GSM SIM card tray

IP Network Connection

  • IPv4 (RFC 791)
  • MAC Address (IEEE 802.3)
  • DMZ and MAC Clone Setting
  • IP/ICMP/ARP/RARP/SNTP
  • Static IP
  • DHCP Client (RFC 2131), WAN
  • DHCP Server, LAN port
  • Wire line speed more than 85MB at Bridge mode
  • PPoE
  • DDNS
  • DMZ
  • VLAN : 802.1Q/1P
  • VPN: PPTP and L2TP without encryption
  • Virtual Server (DHCP Server IP range)
  • DNS Client
  • SNTP support Daylight Saving Time (DST) configuration
  • SNTP with time zone
  • TCP/UDP (RFC 793/768)
  • RTP/RTCP (RFC 1889/1890)
  • IPV4 ICMP (RFC 792)
  • TFTP Client
  • QoS Support : ToS

SIP Protocol

  • RFC3261 compliance
  • SIP Proxy compatible with Asterisk or free download SIP Server
  • SIP UDP Protocol
  • Support SIP compact Form
  • SIP Session Timer (RFC 4028)
  •  MD5 Digest Authentication (RFC2069/RFC2617)
  • Message Waiting Indication (RFC3842)
  • Event Notification (RFC3265)
  • REFER (RFC3515)
  • Support DNS SRV to locate SIP Server (RFC 3263)
  • Support STUN NAT Traversal
  • Support “RPort” parameter (RFC 3581)
  • Support P2P call
  • Only Accept call from registered SIP Server
  • Set up “User Agent” for SIP protocol

Audio Codec

  • G.711 A-law/μ-law, G.729A, GSMFR, G.723.1(6.3K, 5.3K)
  • G.711 A-law/μ-law, G.729, G.723, G.726, GSM codec
  • Silence Suppression
  • VAD/CNG
  • Jitter Buffer : Up to 96 packets
  • Echo Cancellation
  • Packet Loss Compensation
  • Automatic Gain Control
  • In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
  • Adaptive/Configurable Jitter Buffer
  • Acoustic Echo Cancellation

Call Features

  • GSM Call services: Dial, Answer and Hang up
  • Audio Control: Echo Suppression, Noise Suppression, Side
    Tone and Gain Control.
  • GSM Quad Bands 850/900/1800/1900 MHz
  • Operate on any GSM network to provide voice calls over IP
  • Global Country Based Tone Specification
  •  Voice NAT Traversal : Disable, STUN(Type 1,2), STUN (All), enter STUN Server IP address, UPNP
  • Behind NAT (enter IP sharing address)
  • Out-Band DTMF : RFC2833 and SIP
  • RFC2833 Payload type: disable, 101 or 96 
  • DTMF send out ON and OFF Time configure
  • DTMF Detect Minimum ON Time(msec) configure
  • DTMF Detect Minimum OFF Time(msec) configure
  • DTMF Relay Volume configuration
  • Minimum Jitter Buffer(msec) configure
  • Maximum Jitter Buffer(msec) configure
  • Max Echo Tail Length (G.168): 32, 64 and 128ms
  • Keep Alive disable or enable (Time configure in second)
  • Jitter Optimizer Factor configure
  • Call Hold
  • Call Transfer
  •  Tone Generation: Ring, Ring Back, Dial, Busy, call waiting
  • Out-Band DTMF : RFC2833 and SIP Info
  • Music-on-hold support (via IPPBX or local)
  • Peer to Peer call without SIP Server/IP-PBX
  • Call Forward
  • GSM incoming call forward to assigned IP device (SIP number or IP address with port number)
  • SIP IP incoming call to GSM directly (1-stage dial)
  • Support Dial Plan: Add, Drop, and Speed Dial Rule
  • Caller ID

Management

  • Administrative Telnet CLI and HTTP
  • 3 Levels of User Access Right with Password protection
  • Management from WAN enable or disable
  • Provides System Status Logs from webpage

 

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