Name: 2014 Ejointech big promotion at the end of year,VOIP Gateway gsm/cdma/wcdma16 port 64sim voip gateway buy 10 get 1 free
Brand: EJOIN
Availability: InStock

Product Details

Specifications

1.SIP Protocol
2.IMEI change
3.BTS change and lock
4.SIM rotation

Product Description

ACOM516-64/128 VoIP Gateway

ACOM516 series is able to process traditional voice call service and internet data service. Compare to the ACOM508 GSM/CDMA/WCDMA series VoIP Gateway ,ACOM516 series adapt new hardware and software structure, which supports up to 16 concurrent calls.

Special Features:

l 16 channels, up to 128 SIM cards

l GSM, CDMA, WCDMA VoIP gateway

l IMEI change, BTS change and lock,SIM rotation

l SIP protocol, compatible with Asterisk, Elastix, Freeswitch

l SMS batch receiving/sending, AT command compatible

l Support G729a/b/e,G723.1,G.711 A/U law, iLBC auto-selecting

l EBO(Ejoin Bandwidth Optimization)

l Proxy Encryption Solution for IP Block

l Support SIM Bank

l VPN(pptp)

l SIM Card Rotating(ACOM516-128)

u Solution to solve the SIM card blocked by:

² Accumulated Call Duration

² Accumulated Connected Calls

² Accumulated Calls

² Consecutive Failed Calls

² Consecutive No-Alert Calls

² Consecutive No-Answer Calls

² Consecutive No Carrier Calls

² Consecutive Short-Duration Calls

l Station intelligent switching

l ERMS(Ejoin Remote Management System)

l Port Inter-Calling

l Fake Ringback

l Call Duration Limitation for SIM Card/Single

l Dial Plan/Prefix Inward Translation/Intelligent Routing

l Web Browser: Firefox/Chrome /IE/Opera

Specification:

Model	ACOM 516-128
Number of Channels	16 channels
Frequency	GSM: 850/900/1800/1900MHz
	CDMA:450/800/1900MHz
	WCDMA:850/900/1800/1900/2100MHz
SIP Specification	SIP/2.0 RFC3261
	Session Timer RFC4028
	STUN
Network Protocols	DHCP/PPPoE/VPN(pptp)
	NTP
	Telnet/HTTP/FTP/TFTP
	Encryption:Ejoin,VOS2000,RC4,XOP.Base128
Telephony Features	Hot-line call ,Dial plan, Speed dial, Phone book, CDR, LCR, White/Black list
Telephony Signaling	DTMF tone dectection/generation
	DTMF relay: in-band, RFC 2833 and SIP info
	Call forward: unconditional, no answer and busy
	N-way conferencing
	Caller ID display/generation, Mobile Number Portability
Voice Capability	Voice codecs:G729a/b/e,G723.1,G.711 A/U law, iLBC
	Echo cancellation
	Silence suppression & detection(VAD, CNG)
	Adaptive jitter buffer
	Volume adjustable
	IVR customized
Number of Ports	1 WAN 10/100Base-T ethernet(RJ-45 connector)
	1 LAN 10/100Base-T ethernet(RJ-45 connector)
	1 Console(USB)
LEDs	1 Power and 16 groups of card online and running status indicator
Power Supply	100-240VAC, 50 - 60 Hz IN, 12V/5A OUT
Operating Environment	Operating temperature: 0 - 50ºC
Operating Environment	Operating humidity: 10 – 90%RH
Warranty	12 Months

Mobile Features:

l SMS Send, Receive and Forward(GSM/SIP/HTTP)

l SMS Inbox

l AT Command , USSD

l SMS Format: PDU/TXT

l PIN Code Management

l CDMA Delay Answer

l GSM Polarity Reversal

l Carrier Selection

l Caller ID Hidden(need SIM Card support)

Maintenance and Management:

l Multi-language Interface

l USB Serial COM

l Configuration Backup and Restore

l Support HTTP/TFTP Upgrade

l Call statistics: ASR,ACD,PDD

l Ejoin Remote Management System

Detailed Images

Operating principle

Gateway 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.

1.Application

1)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.

2)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.

2.Solution

1)Used in conjunction with VOS, VPS, call back or other Asterisk-based softswitches, for accessing VoIP network by SIM cards, LCR select least cost router to reduce the cost of calls.

2)Saving Cost on International and Long Distance Calls,when you are in the other countries& you have this gateway at your Local office,you could dial & receive phone call from your customers as you are there without Long Distance fee Very excellent solution for internal company with branch offices oversea.

Packaging & Shipping

1. Accesories:

2.package detail:

3.Shipping:

Our Services

Part 1.

> We provide strong tech support about VOIP server and A-Z termination.( 7×24 Hours serve for you !!! )

>We deal with voip minutes and offer voip terminal guide, may need your terminal routes.

>We offer full tech support untill your gateway runs.

Part 2 .

If you have any needs,welcome to contact me :

Name: Cherry Chang

Skype: ejointech-h

QQ Chat:1414209628

Viber/whatsapp: +86 18476572872

Main product catalog

Model	508G-8	508G-32	/508C-8	508C-32	508/516W	508W-32
	/516G-16	/516G-64	/516C-16	/516C-64		/516W-64
	/532G-32	/532G-128
Number of Channels	8/16/32GSM	8/16/32GSM	8/16CDMA	8/16CDMA	8/16WCDMA	8/16WCDMA
Number of SIM cards	8/16/32	32/64/128	8/16	32/64	8/16	32/64
Frequency	850/900/1800/1900MHz		450MHz /800/1900MHz		850/900/1800/1900/2100MHz
Call Control Protocols	SIP
Voice Codecs	G.729A, G.723.1, G.711 a/ulaw