DBL TECHNOLOGY LIMITED
DBL TECHNOLOGY LIMITED
Codec G.711 VoIP GSM Gateway / Remote SIM Bank for Call Ter
  • Codec G.711 VoIP GSM Gateway / Remote SIM Bank for Call Ter
Codec G.711 VoIP GSM Gateway / Remote SIM Bank for Call Ter

Codec G.711 VoIP GSM Gateway / Remote SIM Bank for Call Ter

Min. Order:
1
Min. Order:
1
Delivery Time:
30 Days
Quantity:

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Basic Info
Basic Info
Place of Origin: shenzhen, China
Product Description
Product Description

Codec G.711 VoIP GSM Gateway / Remote SIM Bank for Call Termination

Specification:

  • Brand name: DBL

  • Model Number: GoIP 4

  • Type: VoIP GSM Gateway

  • SIM Cards : 4 sim card

  • Size:25*14*6 cm

  • Protocol: SIP & H.323

  • Support: SMS Server , VPN

Quick detail:

1. Quad band

2. Bulk SMS/USSD

3. IMEI changeable

4. Support SIM Bank

5. VoIP SIP&H323

6. Remote Access

7. Relay and VPN

8. ASR &ACD monitoring

9. Optional SMS termination

10.Auto Balance and Recharge

Competitive Advantage:

  • Remote SIM operation for SIM Card management
  • SMPP support for 3rd party development of SMS Applications.
  • Free server utilities for remote access and SMS management.
  • Telnet mode for sending AT commands to GSM module.
  • Compact and light weight design
  • Direct customer support

The GoIP series gateway is a broadband relay gateway newly developed by DBL Technology. It is a new product for seamless connection between the GSM network and VoIP network. When the mobile phone SIM card is installed in the GoIP, users can register the GSM phone to the VoIP softswitch system. Through the GoIP, users can realize the uplink and downlink calls between the GSM network and the VoIP network. In addition, the GoIP supports the transparent transmission of the caller number from the PSTN to the VoIP.

GoIP is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks. GoIP is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GSM gateway also provides significant savings in usage, infrastructure and maintenance cost compared to conventional PSTN.

The GoIP features embedded SIP and H.323 protocols with flexible setting. The bi-directional password authentication (call authorization) and trust list authentication greatly minimize the risk of charge losses and the flexible routing function can meet special requirements of various call forwarding. In particular, the GoIP gateway supports multi device groups, with flexible setting of large GSM gateway groups with different channel numbers. With its low price, excellent voice quality, and powerful features, the GoIP series gateway is the first choice for system integrators, traffic operators, and softswitch manufacturers.

VoIP GSM Gateway GoIP bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.

Key Features

Open Standard VoIP Protocols (SIP&H.323)

Single or Multiple Server Registrations

Two 10/100 Ethernet for WAN / LAN connections

Peer-to-Peer IP Calls

Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer

Line Echo Cancellation

VLAN and QoS support

NAT Transversal and Router functions

Voice prompts, HTTP Web, Auto Provision support for configuration and updates

Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features

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

Dial in mode or dial out mode only

Call forward from GSM to VoIP and VoIP to GSM

Dial Plan

Retransmit GSM Caller ID to VoIP terminal

Enhanced Features

Dynamic selection of codec

Advanced jitter buffer

Automatic traversal of NAT and firewall

VLAN / Qos

Router

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 - Simple Traversal of User Datagram Protocol (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

Operating temperature: 10°C to 40°C (50°F to 104°F)

Physical and Environmental

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

Power: 12 VDC 2A (110V-220V) (AC/DC adapter included)

Warranty: one year

Applications:

1. Call Forward

1.Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using VoIP.

2.Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network.

3.As shown in the network topology diagram, a VoIP Service Provider is using GoIPs as call origination and termination devices.

4.A call dialed to a GoIP (right hand side) via GSM is first routed via VoIP and then terminated via a VoIP end point or VoIP Service Provider.

5 .A VoIP call originated from the left hand side is routed to a GoIP on the right hand side and then is dialed out as a GSM call.

2) Call Back

1,Call Back is referring to the telecommunications event that occurs when the originator of a call is immediately called back in a second call as a response.

2,GoIP could be used to achieve this function alone or as an terminal that is integrated in an existing call back server / platform.

3,For standalone operation, GoIP receives a call with caller ID information and then rejects the call immediately without answering the call. GoIP then calls back the caller so that he can dial a phone number to make a call. In this case, GoIP must register to a VoIP Service Provider who can offer terminate the call.

4,In a call back system, GoIP acts as a device to initiate the call back function. Typically, this is done in two ways. The first method is to send an SMS with the callee’s phone number to the GoIP. The GoIP then sends both the caller’s and callee’s phone numbers to the call back server to complete the call back function. The second method is to call the GoIP and the hang up (with the call being answered). GoIP sends the caller’s phone number to the call back server and the call back server calls the caller directly so that the caller can then dial a phone number to make a call.

FAQ:
1. Main function of GoIP
For call terminal, call forward/call back
2. Supported Software?
VOS,VPS,VoIP switch,3CX,P·B·Xes,free PBX,MOR
3. Supported hardware
IPPBX and so on
4. Bandwidth requested by GoIP
Up to your codec. G723 23kbps/ line G729 34kbps / line, G711 90 kbps/ line

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