16 port goip gsm voip gateway

VoIP GSM Gateway GoIP-16  GoIP is a VoIP GSM Gateway for call termination (VoIP to GSM) and origination (GSM to VoIP). It is SIP&H.323 based and compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster. It can enable to make 1/4/8/16 calls simultaneously from IP phones to GSM networks and GSM networks to IP phone.

Key Features

16 GSM channels, up to 16 SIM cards

For call termination (VoIP to GSM) and origination (GSM to VoIP)

Standard SIP & H.323 protocol, Communicates with other gateway or PC

Quad band, IMEI changeable, Remote Access

Support of SMB32 SIM Bank

Optional SMS termination

Allows your program send/receive SMS with AT command

Easy to install and administrate

Auto Balance and Recharge

Auto BTS changeable

Support one stage dialing

Support free mode-two stage dialing and assigned mode-one stage dialing

Call Back feature

All functions can be set on web

Provide CDR

Enhanced Features

LEDs for Power, Ready, Status, WAN, PC, GSM

Dial in mode or dial out mode only

Call forward from GSM to VoIP and VoIP to GSM

Dial Plan

Password protection for both GSM dial in or dial out

Retransmit GSM Caller ID to VoIP terminal

Dynamic selection of codec

Advanced jitter buffer

Automatic traversal of NAT and firewall

VLAN / Qos

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 (STUN)- Simple Traversal of 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

Web-base Management

PPP over Ethernet (PPPoE)

PPP Authentication Protocol (PAP)

Internet Control Message Protocol (ICMP)

TFTP Client

Hyper Text Transfer Protocol (HTTP)

Dynamic Host Configuration Protocol (DHCP)

User account authentication using MD5

Free Software

SMS Server

SIM Server ( Sim Bank Scheduler Server )

Relay Server

Remote Access

Technical Specifications

Protocols: SIP/H.323

DTMF sending: REC2833

No. of voice channels: 16

TCP/IP: IP/TCP/UDP/RTP/RTCP/,CMP/ARP/RARP/SNTP,
           DHCP/DNSClient,IEEE802.1P/ Q.ToS / DiffServ,
           NAT Traversal, STUN, uPnP,
           IP Assignment, Static IP, DHCP, PPPoE

VoIP codec: G.711 PCM A-law/u-law (selectable)                                          
                   G.729 Annex CS-ACELP at 8 kb/s
                   G.723.1 MP-MLQ/ACELP at 6,3/5,3 kb/s (optional)

Voice Quality, VAD, CNG, AEC, LEC, Packet loss

Network bands: GSM850/900/1800/1900MHz

Configuration way: WEB interface

Hardware Specifications

Processor: ARM9E 133MHz

DSP: VPDSP101-4 100MHz

Memory: RAM 16MB/ Flash 4MB

GSM Module: 850MHz, 900MHz, 1800MHz, 1900MHz

Power: 12 VDC 4A (110V-220V) (AC/DC adapter included)

Power consumption: 30W maximum

Operating temperature: 10°C to 40°C (32°F to 104°F)

Storage temperature: 0°C to 50°C (32°F to 122°F)

Size: 200mm (W) x 380mm (L) x 85mm (H)

Weight: 3KG (Including AC/DC Adapter)

Warranty: one year