8 Port/Channel GSM VoIP Gateway with SIP&H. 323 (GoIP_8)

  • Price:

    Negotiable

  • minimum:

  • Total supply:

  • Delivery term:

    The date of payment from buyers deliver within days

  • seat:

    Guangdong

  • Validity to:

    Long-term effective

  • Last update:

    2017-11-07 09:04

  • Browse the number:

    205

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Company Profile

Shenzhen DBL Technology Limited

By certification [File Integrity]

Contact:Mr. Sasa(Mr.)  

Email:

Telephone:

Phone:

Area:Guangdong

Address:Guangdong

Website: http://dbltek.rqxiangye.com/

Product details

8 Port/Channel GSM VoIP Gateway with Sip&H. 323 (GOIP_8)

GoIP GSM Gateway Specification:

1. 8GSM channels
2. Bulk SMS/USSD
3. Good ACD,ASR
4. Auto Balance and Recharge
5. VoIP SIP&H323
6. Remote Access
7. Optional SMS termination
8. Support IMEI changeable
9.Quad band, IMEI changeable

Overview

 

A VoIP GSM Gateway (GoIP) bridges voice communications between the GSM and the IP networks seamlessly.In principle, it consists of a GSM module and a VoIP module. With a SIM card installed, the GSM module registers to the GSM network for voice and SMS services.

 

GoIP comes with various models of 1, 4, 8, 16 and 32GSM channels. When integrating in a VoIP system, GoIP is highly scalable in meeting customer's requirement on the number of channels (lines). In addition with its power functions, high voice quality, and low price, GoIP is an inevitable choice of many system integrators, call termination operatiors, small companies, and individuals.

 

Key Functions

  • GSM Carriers and Base Stations Optional
  • Sending SMS / USSD via built-in webpage or SMS Server
  • Remote SIM
  • Auto balance check and recharge (prepaid SIM)
  • Call time restrictions
  • Remote Control Management
  • Relay Server support for NAT and encryption of IP packets
  • PPTP VPN and Relay Encryption
  • Built-in SIP Proxy for peer-to-peer operation
  • GSM Quad Band: 850 / 900/1800/1900 MHz
  • Call forward from GSM to VoIP and VoIP to GSM
  • Call Back

 

Key Advantages:

  • Lightweight and Portable
  • Easy to install and administrate
  • IMEI Changeable
  • GSM Base Station Optional
  • Support SIM Bank/ SIM Sever
  • Manual/ Automatic Selection Operators
  • Sending and Receiving SMS and USSD (Web Interface)
  • Flexible Dial plan fo number alternation, call screen.
  • Flexible configuration modes.
  • All functions can be set on web.
  • Provide CDR report

 

Free Software/Server

Remote Control Server----Access Interface Remotely.

 

Relay Server----Relay Encryption( Make Terminals Traversal the NAT without STUN and Outbound Proxy)

 

SMS Server----Remote SIM cards; Send Bulk SMS; Provide CDR and ASR; Auto Balance and Recharge.

 

SIM Server----Remote SIM Cards on Duty; Set GSM Group( Assign several SIMs per GSM Port); Set talk time per SIM, Set Day of week, time range; Monitor CDR, ASR, ACD.

 

Autocfg Server---- Maintence all DBL termial device anywhere; embeded remote control server; Remote sim auto configuration.

 

Protocols: SIP/H.323

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







Package info


 
 
 
 
 
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 4.5A (110V-220V) (AC/DC adapter included)
Warranty: one year
 
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal