Specifications
ide one (GoIP) or four (GoIP4) cellular channels for IP-PBXSupport open standard SIP Protocols (IETF SIP V2)
Support SIP
16 channels 16 port voip,asterisk gsm gateway GoIP-16 GoIP-16
Specification:
1. 16 GSM channels,up to 16 sim cards
2. Quad band ,IMEI changeable
3. Support of SMB32 SIM Bank
4. VoIP SIP,H323,Remote Access
5. Optional SMS termination
6. Easy to install and administrate
7. Auto Balance and Recharge
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, | |
VoIP codec: G.711 PCM A-law/u-law (selectable) | |
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 | |
