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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.