1.Introduction MV-370/MV-372 is a 1/2 channels VoIP GSM Gateway for call termination (VoIP to GSM ) and origination (GSM to VoIP). It is SIP based and compatible with Asterisk. It can enable to make 1/2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone.
5. MV-370 Panel description 14.5cm 3.9cm 17cm (5.1) (5.2) (5.3) (5.4) (5.6) (5.5) (5.7) 5.1 Antenna:Antenna connector. 5.2 DC 12V:Power socket. 5.3 LAN: Standard RJ-45 socket, connecting to Hub circuit. 5.4 PWR: Power indicator light, red light. Light is on when system’s power supply is normal.
6. MV-372 Panel description 6.5 6.6 6.7 6.8 6.1 Antenna:Antenna connector. 6.2 DC 12V:Power input. 6.3 LAN:LAN port. It also can be DHCP Server. 6.4 WAN: RJ-45 internet connector,standard RJ-45 socket,connect to HUB. 6.5 PWR (Power LED):Light up when power is normal. 6.6 VoIP1:an indicator light of VoIP1 6.7 VoIP2:an indicator light of VoIP2 6.8 LINK Indicator:Light up when network is connected.
7.3 Insert the SIM card from back of the main body. (take the slide off first). Click reset button 3 sec. MV-370/MV-372 will restore default IP. Other setting as usual. 7.5 Connect the power adaptor. The ‘POWER’ LED should be light up.
8.Web Page Setting When the IP setting is done, the operator may setup all the rest parameters via web page. Browse the IP address from Internet Explorer (e.g. http://192.168.0.100)。The following page shows up: Enter the username and password for authentication. (default username=voip, password=1234).
9.System Information. 9.1 When you login the web page, you can see the demo system current system information like firmware version, company… etc in this page. 9.2 Also you can see the function lists in the left side. You can use mouse to click the function you want to set up.
10.1 Mobile TO LAN Settings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN. The MV-370/MV-372 will transfer to the URL according to the caller ID of the Mobile. *CID: (1) may enter the whole number, e.g. 0911111111 (2) only part of the number (prefix) e.g.
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When the caller’s prefix number is 0932,MV-370/MV-372 will connect 0911123456 automaticlly (2) Mobile to Lan: *,* Any caller call the MV-370/MV-372’s sim,MV-370/MV-372 will prompt dial tone.Caller can enter IP or sip extension or phone number. *sip extension or phone number both need to register SIP Proxy Server or Asterisk.
Application: a. Call MV-370/ MV-372 b. MV-370/ MV-372 will detect the phone number is in call back list or not c. If yes, MV-370/ MV-372 will reject the call, and call it back d. You will receive the call from MV-370/ MV-372, and prompt a dial tone...
10.3 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time,MV-370/MV-372 will give priority to Mobile to LAN Speed Dial Settings. *The call will be answered and prompt dial tone again. When the caller may enter the “Num”, system will connect the “URL”...
10.4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE. The MV-370/MV-372 will transfer to the mobile number according to the incoming URL *URL:The IP address of the incoming call.
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(3)When you dial any destination phone number from lan phone,MV-370/MV-372 will connect this call auto. Example of Application: When you call the ch.1 MV-370/MV-372 gsm number,it will provide dial tone and you enter a destination number. Then ch.2 MV-370/MV-372 will dial this number and connect.
11.Mobile 11.1 Mobile Status (1)Network Registration:The telecom carrier which the SIM card been registered. (2)SIM Card ID:SIM card ID. (3)Signal Quality:Signal quality. (4)GSM S/N : IMEI Number (5)Incoming IP:The IP address of the last incoming call from LAN. (6)Incoming IP Name: proxy server name (7)Outgoing IP:The IP address of the last outgoing call to LAN.
Only change “mobile” into “on” or “off”,just click “submit”, no 11.2 Mobile Setting need to click “save change” (10) (11) (12) (13) Mobile 1: (6)Rx VoIP Codec (5) Tx DTMF Mobile 2: (1)VoIP Tx Gain Codec (2) VoIP Rx Gain DTMF (1) VoIP Tx Gain: To adjust the volume of LAN side.
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Mobile 2 use the route table for item 25-49 (5)CODEC Tx Gain: as above (6)CODEC Rx Gain: as above (7) SIP From: Caller ID transfer Tel/User(Standard): If you need to register to Asterisk and proxy server, please choose this option. And how to transfer the caller ID to LAN, please refer 21.How to setup Asterisk to receive Caller ID from MV-374/MV-378 (page 42) MV-374/MV-378 will send the message as follows in the Packet.
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(8)Answer Delay: Delay for incoming call when the ring. (9)Presentation CLIR : If you need to block the Caller Id for call termination, please choose Suppression (10)Mobile PIN Code: If you need to unlock pin code via MV-374/MV-378,you can click “On” and enter pin code. (11)LAN Answer Mode: Answered : when mobile answer,then connect the call Alerted : when the mobile is ringing back tone,then connect the call...
11.3 Mobile / Forward Setting : When the first route are busying, SIP can transfer phone call to another free route. When the device are busying, the phone call can be transfer to another device (external equipments). "Forward Enable" is not motivate on Defualt value. So please, mark "Forward Enable"...
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Name URL:Port 192.168.0.100:5060 Fwd to Mobile1: 192.168.0.100:5062 Fwd to Mobile2: Fwd to External: The Explanation of Picture: Fwd to Mobile1:192.168.0.100 : 5060, it means when 5062 Port are busying, SJ Phone can transfer the call to 5060 Port (192.168.0.100). Fwd to Mobile2:192.168.0.100 : 5062, it means when 5060 Port are busying, SJ Phone can transfer the call to 5062 Port (192.168.0.100).
11.4 Mobile / SMS Agent: Read received SMS 2 mode: ASC7(ASCII 7 bit) UCS2(Unicode 16 (1) Rx List: Read received SMS (2) Dest Num: the Receiver’s phone number (3) Message: Please fill the message that want to send to receiver. When you click Rx List, you can view all received SMS as follows.
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Click the serial no,you can view message as follows. -21-...
11.5 Send Bulk of SMS via Microsoft Excel First of all, please open a new Excel file. Step 1 Format Cells Here, we need you to format cells to “Text” first. Please click mouse right key, and choose “Format Cells” Blank A -22-...
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Blank B Step 2 In the Format Cells, please select “Text” Please do this action for Blank A and B both. -23-...
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Step 3 Blank A: is for you to key “phone numbers” Blank B: is for you to key “text” Step 4 Save the file -24-...
11.6 use AT Command via Telnet or your program Allows your program or Telnet Send/receive SMS with AT Command Port : 23 Please enter account and password Choose module Enter “ate1”,then you can see your at command below Enter at+cmgs=”phone number” Enter short message -32-...
12.Network In Network you can check the Network status, configure the WLAN Settings , LAN Setting and SNTP settings. 12.1 Network Status: You can check the current Network setting in this page. 12.2 WAN Settings: You can check the current Network setting in this page.
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12.3 LAN Settings: You can check the current Network setting in this page. (1) The TCP/IP Configuration item is to setup the WAN port’s network environment. You may refer to your current network environment to configure the system properly. (2)DHCP Server: You may refer to your current network environment to configure the system properly -34-...
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12.4 SNTP Settings: SNTP Setting function: you can setup the primary and second SNTP Server IP Address, to get the date/time information. Also you can base on your location to set the Time Zone, and how long need to synchronize again.
13.SIP Setting In SIP Setting you can setup the Service Domain,Port Settings,Codec Settings,RTP setting,RPort Setting and Other SettingS. If the VoIP service is provided by ISP,you need to setup the related informations correctly then you can register to SIP Proxy Server correctly. 13.1 In Servcie Domain Function you need to input the account and the related informations in this page,please refer to your ISP Provider.
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Example: Register VoipBuster Your Voipbuster username Your Voipbuster password Proxy Server’s IP or domain name -38-...
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13.2 Port Setting You can setup the SIP and RTP port number in this page. Each ISP provider will have different SIP/RTPport setting, please refer to the ISP to setup the port number correctly. When you finished the setting, please click the Submit button.
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13.3 Codec Settings: You can setup the Codec priority, RTP packet length in this page. You need to follow the ISP suggestion to setup these items. When you finished the setting, please click the Submit button. -40-...
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13.4 Codec ID Setting You can setup the Codec ID in this page. -41-...
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13.5 DTMF Setting You can setup the DTMF Setting in this page. -42-...
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13.6 RPort Function: You can setup the RPort Enable/Disable in this page. To change this setting, please following your ISP information. When you finished the setting, please click the Submit button. -43-...
13.7 SIP Responses Dial Peer Configuration Table corresponding IP (please read next page) *** If you have dial peer server, Sip server/Asterisk set GSM route,please set Dial Peer server’s IP** 13.7.1 486(busy here), 503(Service unavailable): When Device is busy, you can select 486 or 505 to response to SIP. 13.7.2 180 Ring on/off: LAN TO MOBILE two stage dialing can be turn off, therefore there will be no the Ring Back Tone, all the phone call will be transferred to prompt...
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Edit DialPeer.ini [Window] 14=5070 Xpos=512 15=5072 The second Ypos=252 16=5074 MV-378 Width=471 Total ip / port [RtpPort] Height=399 1=60000 [Info] 2=60002 Total=16 3=60004 [VoipIP] 4=60006 1=192.168.0.100 The first 5=60008 2=192.168.0.100 6=60010 MV-378 3=192.168.0.100 The first 7=60012 4=192.168.0.100 8=60014 MV-378 5=192.168.0.100 9=60000 6=192.168.0.100 10=60002...
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Status The first MV-378 doesn’t register dial peer software The 2,3ch of Second MV-378 idle The 1,4-8ch of Second MV-378 turn off -47-...
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13.8 Other Settings Other Settings: you can setup the Hold by RFC and QoS in this page. To change these settings. please following your ISP information. When you finished the setting, please click the Submit button. The QoS setting is to set the voice packets’...
14. NAT Trans In NAT Trans. you can setup STUN and uPnP function. These functions can help your VoIP device working properly behind NAT. 14.1 STUN Setting: you can setup the STUN Enable/Disable and STUN Server IP address in this page. This function can help your VoIP device working properly behind NAT.
16.Save Change In Save Change you can save the changes you have done. If you want to use new setting in the VoIP system, You have to click the Save button. After you click the Save button, the system will automatically restart and the new setting will effect.
17.Update In Update you can update the system’s firmware to the new one or do the factory reset to let the system back to default setting. 17.1 Update firmware (1) In New Firmware function you can update new firmware via HTTP in this page.
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17.2 Restore Default Settings In this page: Update/ Default Settings, you could restore the factory default settings to the system. All setting will restore default setting. IP will retain original IP as usual not default IP. -53-...
18.Reboot Reboot function you can restart the system. If you want to restart the system, you can just click the Reboor button, then the system will automatically. -54-...
19. IP Setting The operator can setup or query the network parameters by dialing in the mobile number which it SIM card has been put in the main body. The status or result is response by voice. In the first 20 seconds after power-on, the VoIP GSM Gateway enters the IP setting mode.
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IVR will announce the current Check Primary #125# setting in the Primary DNS DNS Server field. Default : 192.168.0.1 IVR will announce the version Check Firmware #128# of the firmware running Version The system will change to DHCP #111# DHCP client Client type DHCP will be disabled and...
CDMA 2000(800MHZ/1900MHZ) **Please note** Most CDMA operators don't offer Polarity reversing . So VoIP to Mobile, MV-370 will connect soon. CDMA operators will start billing soon. It doesn't wait mobile side answer. CDMA Version doesn't support SMS Feature and 180/183...
(1) *,* --->it is two stage dialing. when mobile call in,MV-370/MV-372 will provide dial tone and you can enter ip or asterisk extension or phone number. * If you want to enter phone number,please note your asterisk need to have route of destination number.
22. Appendix: Setup MV-370/MV-372 with Asterisk MV-370/372 Settings Asterisk want to transfer CLID,please choose Tel/Tel (Not Reg) Can register Asterisk or not -60-...
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Set your Asterisk IP,extension or * As Asterisk GSM Route -61-...
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Port Settings: Mobile1 > Sip port: 5060 Mobile2 > Sip port: 5062 Important!!! Don't forget to Save changes and then reboot -62-...
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Asterisk / Trixbox setting Add SIP Trunk: MV-372 must create 2 trunk. First trunk: MV-372 ip:5060 Second Trunk:MV-372 ip:5062 Type your mobile number -63-...
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Type a trunk name Type MV-37X’s ip and port The channel 2 is 5062 port -64-...
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Set GSM Route that dial out via MV-37X After change , please press “Sumbit changes” and “apply configuration changes” The prefix of route : 09XXXXXXXX Choose SIP Trunk -65-...
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15.21 Federal Communications Commission (FCC) Statement You are cautioned that changes or modifications not expressly approved by the part responsible for compliance could void the user’s authority to operate the equipment. 15.105(b) Federal Communications Commission (FCC) Statement This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC rules.
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FCC RF Radiation Exposure Statement: This Transmitter must not be co-located or operating in conjunction with any other antenna or transmitter. This equipment complies with FCC RF radiation exposure limits set forth for an uncontrolled environment. This equipment should be installed and operated with a minimum distance of 20 centimeters between the radiator and your body.
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