Model: | Goip 8 |
---|---|
Brand: | skyline |
Origin: | Made In China |
Category: | Electronics & Electricity / Telecommunication & Broadcasting / Network Communications Equipment |
Label: | gsm gateway , 8 port gateway , voip gateway |
Price: |
US $610
/ pc
|
Min. Order: | 1 pc |
Last Online:29 Aug, 2014 |
IMEI change sip 8 ports gsm gateway
Product Overview
Note: We have A to Z internation routes with best quality and rate! currently Pakisitan Non-Cli routes 92 with Best rates. any inquiries, contact me freely.
if clients buy GOIP gateway from us, we can purchase all of our routes in PK.
The 8-port GSM Gateway (GoIP8) provides the extension of GoIP1 and supports up to 8 GSM channels. Each GSM channel can be programmed individually to establish the connectivity with IP-PBX, or can be bundled as a single trunk in connection with IP-PBX. It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. The GoIP8 brings mobility, scalability and significant savings on long distance charges.
key features
Provide 8 cellular channels for IP-PBX
Support open standard SIP Protocols (IETF SIP V2)
Support SIP proxy mode
Multiple GoIP8 grouping mode
Two 10/100 Ethernet ports for the LAN and an additional device
Guad band GSM module: support GSM 850MHz, 900 MHz, 1800 MHz, 1900MHz
Speech quality ensured by QoS at LAN, IP layers and comprehensive jitter buffer
VLAN and QoS support
NAT Transversal
Voice prompts, HTTP Web
Highly stable embedded Linux operating system in high performance ARM 9 Processor
Enhanced Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Call forward from GSM to VoIP and VoIP to GSM
Dial in mode or dial out mode only
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codecs
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)
Firmware upgrade from GUI
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 User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3892 - SIP Referred-By Mechanism
Proprietary Firewall-Pass-Through Technology
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)
Domain Name System (DNS)
User account authentication using MD5
Hardware Specifications
Processor: ARM9E 133MHz
DSP: VPDSP101-4 100MHz
Memory: RAM 16MB/ Flash 4MB
GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz
Power: Input AC100V ~ 240V, output 12 Vdc 2000 mA
Power consumption: 5W 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)
Working Humidity: 40% ~ 90% Not congealed
Weight: 1.2k g (3 lb) (Including AC/DC Adapter)
Warranty: 1 year
Common function:
1 PSTN to VoIP
Description: using the GoIP to connect with VoIP
2 VoIP to PSTN
Description: using the GoIP to connect with PSTN
3 Calling forward
Description:If you are in china but your main business is in Malaysia, you only put a Malaysia SIM card into the GoIP.In this condition,all calling the SIM number can connect your Phone number in China directly.
4 calling back
Description:When you are using the telephone, and you are want to get preferential from VoIP Phone anytime and anywhere.You just call the SIM card number which in GoIP,the calling number would send to Server by GoIP,then the server receive the calling number and establish the new calling.Then you will receive the new calling,you just accept the calling,Now you are calling with your customer by server.
NOTES
>We provide strong tech support about VOIP server and A-Z termination. (6 days*12 hours)
>We can help you rent a VoIP server in USA.
>We can provide copy version VOS.
>We can install Relay server, SMS server,SMB server for you.
(for free)
>We can offer voip terminal guide.
>We may need your termina routes.
>We offer full tech support untill your gateway runs.