MV-3716 / MV-3732
VoIP GSM Gateway
User Manual
MV-3716
MV-3732
PORTech Communications Inc.
【Content】
1. Introduction........................................................................................................... 1
2. Function description............................................................................................ 1
3. Parts list ................................................................................................................ 1
4. Dimension: 37*26*10 cm...................................................................................... 2
5. Chart of the device ............................................................................................... 3
6. Web Page Setting ................................................................................................. 5
7. System Information.............................................................................................. 6
8. Dial Peer................................................................................................................ 7
8.1 Status.............................................................................................................................. 7
8.2 Settings...........................................................................................................................9
8.3 Prefix.............................................................................................................................13
8.4 Call Data to Server (CDR).........................................................................................17
9. Route ................................................................................................................... 21
9.1 Mobile TO LAN Settings............................................................................................ 21
9.2 Mobile to LAN Speed Dial Settings.......................................................................... 24
9.3 LAN to Mobile Settings...............................................................................................25
10. Mobile................................................................................................................ 28
10.1 Mobile Status.............................................................................................................28
10.2 Mobile Setting............................................................................................................ 29
10.3 Mobile / SMS Agent:.................................................................................................33
10.4 Send Bulk of SMS via Microsoft Excel.................................................................. 35
10.5 Use AT Command via Telnet or your program.................................................... 43
10.6 USSD SIM Balance Check via Telnet................................................................... 45
10.7 SIM Setting................................................................................................................47
10.8 Operator Setting........................................................................................................ 49
10.9 BCCH Info.................................................................................................................. 51
10.10 USSD (Unstructured Supplementary Service Data).........................................56
11. Network ............................................................................................................. 65
11.1 WAN Setting.............................................................................................................. 65
11.2 SNTP Settings...........................................................................................................66
12. SIP Setting......................................................................................................... 67
12.1 Service Domain Setting........................................................................................... 67
12.2 Ports Setting.............................................................................................................. 69
12.3 Codec Settings:.........................................................................................................70
12.4 Codec ID Setting.......................................................................................................71
12.5 DTMF Setting............................................................................................................72
12.6 SIP Responses ......................................................................................................... 73
12.7 Other Settings ........................................................................................................... 75
13. STUN Setting..................................................................................................... 76
14. Update ............................................................................................................... 77
14.1 Update Firmware......................................................................................................77
14.2 Default Settings.........................................................................................................78
15. System Authority.............................................................................................. 79
16. Save Change..................................................................................................... 80
17. Reboot............................................................................................................... 81
18. Specification ..................................................................................................... 82
18.1 Protocols....................................................................................................................82
18.2 TCP/IP........................................................................................................................ 82
18.3 Codec ......................................................................................................................... 82
18.4 Voice Quality ............................................................................................................. 82
18.5 GSM (MV-3716/MV-3732).......................................................................................83
19. Simple Steps..................................................................................................... 84
20. Appendix: Setup MV-37x with Asterisk .......................................................... 85
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1. Introduction
MV-3716/MV-3732 is a 16 / 32 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 16 / 32 calls
simultaneously from IP phones to GSM networks and GSM network to IP
phone.
2. Function description
2.1 VoIP(SIP)、GSM conversion.
2.2 50 sets of LAN->MOBILE routes setting，50 sets of MOBILE->LAN
routes setting.
2.3 Voice response for setting and status (dial in from mobile).
2.4 Series connections to save bills.
2.5 Standard SIP(RFC2543,RFC3261) protocol，
*It communicates with other gateway or PC.
3. Parts list
3.1 「MV-3716/MV-3732」main body
3.2 Power adaptor
Output 12V/9A, Input 100~240V Auto switching
3.3 Network cable
3.4 Antenna: MV-3716: 4 pcs / MV-3732: 8 pcs
3.5 Rack-mount accessories (compatible with 19“Rack)
3.6 User Manual
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(3.5)
4. Dimension: 37*26*10 cm
(3.1) MV-3716
(3.4)
(3.3)
(3.1) MV-3732
(3.2) Power adapter
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5. Chart of the device
1 Antenna：Antenna Connector
2. SIM Holder: Insert the SIM card as instruction and hear click sound
(the chip side down); Press the SIM to bottom with click sound to
remove the SIM card
3. PWR (Power LED)：Light up when power is normal.
1. DC 12V：Power input.
2. WAN: RJ-45 internet connector
3. LAN: For maintenance use, not for any propose
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1. Dial Peer Reset Button
2. IP Reset Button:
Press this button about 10 seconds
IP restore back to 192.168.0.100
3. DHCP mode Button:
Press this button about 10 seconds and switch to DHCP mode
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6. 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). The page follows when the
username and password are correct.
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7. System Information
User can see the demo system current system information like firmware
version, company… etc in this page.
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8. Dial Peer
8.1 Status
1. ch: The port of GSM channel
2. grp: the group of GSM channel
3. state:
INIT/0: GSM module is initialing
IDLE/0: GSM module not register
IDLE/1: GSM module registered
M.ringback/0: Ring Back
M.dialed/0: GSM port is dialed
M.listen/0: GSM port is engaged
-8-
4. MNC: Mobile Network Code
5. SQ: Signal quality
6. Mobile: The caller number of the incoming/outgoing call to Mobile
7. dir: The Arrow shows the route to be LAN to Mobile or Mobile to LAN
a. < : LAN to Mobile
b. > : Mobile to LAN
8. LAN: the IP address of the last incoming/outgoing call from/to LAN
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8.2 Settings
1. Transfer SIP Message
The Replace contact to dial peer: The default is OFF, which won’t send
the SIP message to corresponding port through Dial Peer.
If ON, all SIP messages will send to corresponding port via Dial Peer.
2. SIP Response when all busy
User can select the corresponding response while all ports are busy.
The Default is 600
600 : Busy Everywhere (default)
408 : Request Timeout
480 : Temporarily unavailable
503: Service unavailable
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3. Dial Peer
Working Mode
a. OFF: To disable Dial Peer, user need to assign the port of GSM channel for the
incoming calls from LAN side (E.g. Default ch1 is 5064 port; ch2 should be 5066
port and so on)
b. Internal: to motivate Dial Peer, all incoming calls from LAN will come to dial peer
port. Dial peer will route calls to idle channels(Default: 5060 port)
c. External: All GSM Channel are controlled by external Dial peer program.
External URL External Dial peer program’s IP address and port number
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Edit DialPeer.ini (External Dial Peer)
[Window]
Xpos=512
Ypos=252
Width=471
Height=399
[Info]
Total=16
[VoipIP]
1=192.168.0.100
2=192.168.0.100
3=192.168.0.100
4=192.168.0.100
5=192.168.0.100
6=192.168.0.100
7=192.168.0.100
8=192.168.0.100
9=192.168.0.110
10=192.168.0.110
11=192.168.0.110
12=192.168.0.110
13=192.168.0.110
14=192.168.0.110
15=192.168.0.110
16=192.168.0.110
[SipPort]
1=5060
2=5062
3=5064
4=5066
5=5068
6=5070
7=5072
8=5074
9=5060
10=5062
11=5064
12=5066
13=5068
14=5070
15=5072
16=5074
[RtpPort]
1=60000
2=60002
3=60004
4=60006
5=60008
6=60010
7=60012
8=60014
9=60000
10=60002
11=60004
12=60006
13=60008
14=60010
15=60012
16=60014
[PtcPort]
1=40000
2=40000
3=40008
4=40008
5=40016
6=40016
7=40024
8=40024
9=40000
10=40000
11=40008
12=40008
13=40016
14=40016
15=40024
16=40024
Total ip / port
The first
MV-378
The second
MV-378
The first
MV-378
The second
MV-378
The second
MV-378
The first
MV-378
The second
MV-378
The first
MV-378
The second
MV-378
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External Dial Peer Log
You can check the Statue here
1. CH: The number for GSM port of MV-37X
2. MvIP: The IP address of MV-37X for Dial Peer connection
3. Port: The corresponding port for MV-37X
4. Sq: Signal Quality for MV-37X GSM Port:
5. State: The GSM Port Sate status
INIT/1: GSM module is initialing
IDLE/0: GSM module is not register
IDLE/1: GSM module is registered
BUSY: GSM Port is busy
LISTEN: GSM port is engaged
OFF/0: GSM module is out of working
6. Remote: The VoIP Sender’s IP
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8.3 Prefix
User can setup the prefix number in 15 groups. Dial peer will route the
calls based on the prefix settings of each group
1.Group Enable
Off: The default is off.
On: Dial peer will route the calls based on the prefix settings of
each group. And Dial Peer status will show the grp
information as below.
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Please click to select the group number of each channel
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After setting, please click submit button
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2.Group: The group number, total is 15 sets
3. Name: Fill the name of the group
4.Prefixs: Fill the local area number or prefix numbers of the group
After all settings are done, please click submit button.
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8.4 Call Data to Server (CDR)
It can provide Call Detail Record (CDR) for traffic and accounting
management. User need to download external Dial Peer software on PC
and can monitor traffic.
1.Data ID: MV will create one default Data ID
2.Server URL: Fill the IP and port of the CDR server
After the setting, please click Submit and save change button to wait for
system reboot
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External Dial Peer
You can check CDR Statue here
1. ID: The MV’s Data ID
2. CH: The GSM channel of MV-37X
3. Cimi: The SIM Card ID
4. LAN: Show the outgoing LAN IP or Incoming LAN IP
5. Dir: The Arrow shows the route to be LAN to Mobile or Mobile to LAN
6. Mobile: The outgoing mobile number or incoming mobile number
7. tStart: When the call started(date and time)
8. tANS: The second answering the call
9. tEND: The second ending the call(duration)
(tANS, tEND are the exactly talking seconds)
10.State: The GSM Port Sate status
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CDR Files store at C:\Program Files\DialPeer
The CDR log is stored in this “cdr” file each hour, which includes all gsm port call
detials record.
If there’s no calls in this hour, it won’t creat any log.
CDR File
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Example:
1. Id=Mv-000000: The MV’s Data ID
2. Ch=1: The 1stchannel for MV ID
3. Cimi=466922202956645 : The SIM card ID for this GSM port
4. dir=L2M: The route is LAN to Mobile (If it’s Mobile to LAN, that shows M2L)
5. iurl=192.168.0.96: The incoming IP
6. omb=0980763178: The outgoing number
7. tStart=4e7a0682(2011/09/21 15:45:06): The duration for the call
8. tEnd=+26: The call end on 26thsecond
9. state=LANEnd: The call hang up on LAN side.
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9. Route
9.1 Mobile TO LAN Settings
User can assign the routing rule to transfer the call incoming on
MOBILE to LAN
Please move the mouse to that red arrow spot and click
It will show the setting bLANk. After the setting, please click Entry.
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1. MCH: the code of mobile channel
2. CID:
(1) It may enter the whole number, e.g. 0911111111
(2) Only part of the number (prefix) e.g. 0911* means any number
starting with 0911 will be accepted
(3) * means all numbers can be accepted
Please note the priority of the rules. The item which has more digits will
have higher priority. If the digits are the same, then former one gets the
higher priority.
3. URL：The IP address to transfer this call
(1)It may enter the whole IP address, e.g. 192.168.0.101 or proxy
extension or phone number.
(2)If an ‘*’ entered, it means 2-stages-dialing. The call will be answered
and prompt dial tone again to receive the IP address/sip extension
or any phone number as the destination. The caller may enter the
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IP such as 192*168*0*101#.
*If the device have register proxy server/Asterisk ,you can enter any
destination phone number. Please note the proxy server/Asterisk
need to set the route of destination phone number.
4. SEL: Select the one to delete
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9.2 Mobile to LAN Speed Dial Settings
NOTE: It’s for 2 stage dialing mode
The call will be answered and prompt dial tone again. When the caller
may enter the “Num”, system will connect the “URL” as destination.
E.g. item: 0 Name: JACK URL: 192.168.0.156,
When the caller hear dial tone and enter 0, system will connect
192.168.0.156
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9.3 LAN to Mobile Settings
User can assign 24 sets of routing rule to transfer the call incoming
from LAN to MOBILE. The chart setting is used for all channels.
Please move the mouse to that red arrow spot and click
It will show the setting bLANk. After the setting, please click Entry.
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1. No. : The code number
2. URL: It’s the IP address of the incoming call
It may enter the whole IP address, e.g. 192.168.0.101 or proxy server’s
extension. If a simple ‘*’ is entered, means no restriction for the
incoming IP address.
3. Call Num:
(1). May enter the whole number, e.g. 0911111111
(2). A simple *”means 2-stages-dialing. The call will be answered and
prompt dial tone again to receive the called number as the
destination, e.g. 0911111111 or 0911111111#
(3). # for one-stage dialing
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(4). # ['d'n]['a'ppp] for one-stage-dialing
[...] is option
'd'n means to delete the beginning n codes,
'a'ppp means to add 'ppp' in front.
For example #d123a456 means one-stage dialing,
delete the first 123 from your destination number,
then add 456 in front as the new destination number.
Example:
LAN to Mobile: *, #
(1)MV-3716/MV-3732 andLAN Phone both need to register proxy server
or Asterisk.
(2)Proxy server/asterisk set the route that the prefix of destination
number
(3)When you dial any destination phone number from LAN phone,
MV-3716/MV-3732 will connect this call auto.
4.SEL : Select the one to delete
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10. Mobile
10.1 Mobile Status
(1)Choose Mobile 1,2,3 or 4 (MV-3732: Mobile 1,2,3,4,5,6,7,8)
(2)Network Registration：The telecom carrier, which is the SIM card been
registered.
(3)SIM Card ID：SIM card ID. (4)Signal Quality：Signal quality.
(5)GSM S/N: IMEI Number
(6)Motion State: The status of SIM card
(7)Incoming IP：The IP address of the last incoming call from LAN.
(8)Incoming IP Name: proxy server name
(9)Outgoing IP：The IP address of the last outgoing call to LAN.
(10)Incoming Mob：The caller ID of the last incoming call from MOBILE.
(11)Outgoing Mob: The called number of the outgoing call to MOBILE.
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10.2 Mobile Setting
(1) VoIP Tx Gain: To adjust the volume of LAN side.
(2) VoIP Rx Gain: To adjust the volume of Mobile side.
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(3)LAN Dial tone Gain: To adjust dial tone gain down of LAN.
(4)Routing Range: The route table -50 sets can share by two
channels(1,2 ch / 3,4 ch / 5,6 ch / 7,8 ch )
ex: Mobile 1 use the route table for item 0-24,
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-3716/MV-3732 (page 42)
MV-3716/MV-3732 will send the message as follows in the
Packet.
From: “caller number” <sip:3001@192.168.0.228>;tag=51088abb
User/User (Standard): If you need to register to Asterisk and proxy
server, please choose this option.
MV-3716/MV-3732 will send the message as follows in the
Packet.
From: " 3001" <sip:3001@192.168.0.228>;tag=51088abb
Tel/Tel :
MV-3716/MV-3732 will send the message as follows in the
Packet.
From: "caller number" <sip: caller number @192.168.0.228>;tag=6ac93f7c
※Please note: If you choose this option, please don’t register to
Asterisk and proxy server. Please only fill proxy server IP and
choose Active: on (else field empty) in sip setting/service domain
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User/Tel
MV-3716/MV-3732 will send the message as follows in the Packet.
From: “Username” <sip: caller number @192.168.0.228>;tag=7f130947
※ If you choose this option, please don’t register to Asterisk and
proxy server. Please only fill proxy server ip,Username and
choose Active: on (else field empty) in sip setting/service
domain
(8)Answer Delay: Delay for incoming call when the ring.
(9)Presentation CLID: If you need to block the Caller Id for call
termination, please choose Suppression
(10) Restart Dial Fail: In this feature, user can initialize and register the
module while GSM module dials fail in couple times. When GSM
module is dysfunctional, it can avoid the device shut down in
advance.
(11)Mobile PIN Code: If you need to unlock pin code via
MV-3716/MV-3732, you can click “On” and enter pin code.
(12) Dial Prefix: The prefix number of outgoing calls. When LAN to
Mobile, MV-3716/MV-3732 will automatically add the“Dial prefix” for
outgoing mobile.
(13)LAN Answer Mode:
Answered: when mobile answer, and then connect the call
Alerted: when the mobile is ringing back tone, then connect the
call
Income: when LAN dial out, then connect soon
(14) Init AT Cmd: User can fill the AT Command for GSM module
(15) Band Type: You can manual setting according to your GSM
Frequency of carrier.
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(16) ON/Off: If you use this channel, please click on. Otherwise, please
click off.
After the setting, please click Submit and save change button to wait for
system reboot
You can click Submit All to copy to Mobile setting, and select Yes and
save change to wait for the system reboot
Please check below:
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10.3 Mobile / SMS Agent:
1. Port: The GSM Channel No.
2. Status:
a. Standby: The GSM Channel is ready and idle for SMS sending
b. Not Ready: The GSM Channel is not registered or engaged, not
able to send SMS
3. Encode : ASC7(ASCII 7 bit) or UCS2(Unicode 16 bit)
4. Via : To select the GSM Channel for SMS sending
5. Dest Num: the Receiver’s phone number
6. Message: Please fill the message that wants to send to receiver.
After typing the SMS, please click Send Now button
Read received SMS
2 mode:
ASC7(ASCII 7 bit)
UCS2(Unicode 16 bit)
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When you click Rx List, you can view all received SMS as follows.
Click the serial no, you can view message as follows.
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10.4 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
-36-
BLANk B
Step 2
In the Format Cells, please select “Text”
Please do this action for BLANk A and B both.
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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
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Save the type as “Unicode Text”
Step 5
Open MVsms_exe - MV-SMS (Configuration Settings)
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Step 6
Please do the configuration as following:
MV-3732
MV-3716
=2
-40-
MV-372 & MV-370
Step 7
Run MV-SMS program
=1
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Step 8
1. Open File
2. Open the “Excel file” that you just saved
-42-
Step 9
Sending
Step 10
Send SMS Complete
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10.5 Use AT Command via Telnet or your program
Allows your program or Telnet Send/receive SMS with AT Command
Telnet PORT Corresponding port as follows:
(2 modules in one SLAVE)
SLAVE 1:1301
SLAVE 2:1302
SLAVE 3:1303
SLAVE 4:1304
SLAVE 5:1305
SLAVE 6:1306
SLAVE 7:1307
SLAVE 8:1308….. *MV-3716
SLAVE 9:1309
SLAVE 10:1310
SLAVE 11:1311
SLAVE 12:1312
SLAVE 13:1313
SLAVE 14:1314
SLAVE 15:1315
SLAVE 16:1316….. *MV-3732
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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 and ctrl+Z
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10.6 USSD SIM Balance Check via Telnet
1. USSD Request: Please enter USSD code for your operator to check
Balance
2. Module command:
Please enter “15” for Siemens BG2W module
Please enter “0” for Simcom module
User can check this information on main page on Module
Description
1. USSD Request
2. Module command
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After sending the USSD request, MV will receive the SMS from operator
Please check the incoming SMS on SMS Agent
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10.7 SIM Setting
1. CU ID: It’s the ID for MV and SIM Server Transfer Protocol, within
1~9999. Each MV under same SIM Sever should setup different CU
ID, and no reusing parameter. E.g. If you put “888” on 1stMV-3732
that you can’t use “888” on 2ndMV-3732, and so on.
2. Mode
a. Local: Disable Remote SIM feature
b. Bank: Enable Remote SIM Bank feature, and manage SIM card on
SBK-32 SIM Bank.
c. Server: Enable Remote SIM Server feature, and allocate SIM
cards on SBK-32 SIM Bank.
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3. Mobile
a. ID: Put in 8 digits (hexadecimal, also base 16), which used for
GSM Module ID identification to Remote SIM protocol. User can
define the ID. IF it’s Server Mode, just leave it default. If it’s Bank
Mode, No reusing GSM Module ID for same SIM Bank.
b. Group: Fill in SIM Group number for Remote GSM module. Server
follow SIM Group Number to allocate SIM card to correspond GSM
module
4. Card ID: Put in 8 digits (hexadecimal, also base 16), which used for
SIM Card ID identification to Remote SIM protocol. User can define
the ID. If it’s in Server Mode, Card ID can be bLANk or default. As for
Bank Mode, Card ID must be corresponding to SIM Card ID of SIM
Bank.
5. Bank URL: If it’s Bank Mode, please fill SIM Bank IP and Port Number.
On other hand, please leave bLANk for Server Mode.
6. Server URL: If it’s Server Mode, please fill SIM Server IP and Port
Number. On other hand, please leave bLANk for Bank Mode.
7. Status: User can check the SIM Card ID of GSM module and IP, Port
Number of SIM bank.
After the setting, please click submit and save change button and wait for
system reboot
-49-
10.8 Operator Setting
1. Operator ID: When GSM module is registered, user can click the List
to show all available operators in that area. You will see like follows
diagram.
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2. Work Mode:
a.Every time reset module:
Fill the assigned Operator ID, then press Submit bottom and save
change. After reboot, GSM module will research the operator ID and
registered the base station.
b.Manual:
Fill the assigned Operator ID, then press Now bottom. GSM module
will search that Operator ID and registered after reboot.
After the setting, please click submit and save change button and wait for
system reboot
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10.9 BCCH Info
Please work with this feature when the mobile status is “Stand by/Active”.
It detects the surrounding active cell, up to 7 cells and shows Cell ID,
signal and best signal (RXlev). The No.0 shows the data of current
registered cell. Follow by No.1 to No.6 cell is based on cell signal (best to
low).
NOTE: Support Quad band-BG2W, Quad band-M10 and firmware V10.185 above
only.
MCC : Mobile Country Code
LAC : Location Area Code
Cell : Cell Identifier
BSIC: Base Station Identity Code
BCCH: Broadcast Control Channel
RxLev: Received Signal level in dbm
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How to Configure
1. You can choose a BCCH channel by clicking on the cell. The module will
automatically register in the new BCCH.
E.g. If you would like to register BCCH channel on No.4 cell, please click no4 select
like below.
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2. System will show the cell number information once you select on Preferred this
Cell form. Please click the submit button and Save Change, and wait for system
reboot
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After system restart and turn to Standby, please check on No.0 cell and confirm the
current registered cell you selected. At the point, the GSM module won’t provide the
data of surrounding cell signal, but shows -110dbm on No.1 to No.6 RxLev, which
means GSM signal 0.
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3. If you would like to research all the surrounding BCCH cells again, please cancel
Preferred this Cell selection first and send Submit, Save Change to restart the
gateway. That, System can detect the surrounding active cell, up to 6 cells and
display Cell ID, signal and best signal (RXlev).
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10.10 USSD (Unstructured Supplementary Service Data)
User can check USSD screen for SIM Balance remaining and SIM recharge (add
value) automatically. Please work with this feature when the mobile status is “Stand
by/Active”. And ensure your Service provider has given you a USSD
string(Command) for checking SIM Balance and Recharge the SIM Card.
1. Balance (SIM Balance remaining)
Step1: Select “Balance”
Step 2: Enter USSD command in Balance column
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Step 3: Enter “entry” button to save the command
Step 4: For Mutli Channel selection in one time, please click the “SEL” on the right
button
-58-
Step 5: Enter “Send Selected” for reply
-59-
System will display the reply on “Response “column
(And system will auto refresh the page in 5 seconds for some late data coming)
-60-
2. Recharge (add value)
Step1: Select “Recharge”
Step 2: Enter USSD command in Recharge column
Step 3: Enter “entry” button to save the command
Step 4: For Mutli Channel selection in one time, please click the “SEL” on the right
button
Step 5: Enter “Send Selected” for reply
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4. Rx Decoder
a. None: GSM Format (Default)
b. ASC7: ASCII 7bit
c.UCS2: Unicode 16bit
When user select default GSM Format(None), it may not receive correct GSM code
due to the different operator or GSM module/chipset. Please check below example,
In this case, user need to select other RX Decoder (ASCII or UCS2) to receive
correct message.
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For Example,
None format: When user send command, “*145*11#”, the return message show on
system, “55E7D2F9BC3A41412894991C06A9C9A713”
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ASC7 Format: In this format, the return message is “UNKNOWN aPPLICAT]ON”
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UCS2 Format: In this format, the return message is “嗧틹밺䅁⢔餜ک즧”
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11. Network
User can check the Network status and configure the WLAN Settings
and SNTP settings.
11.1 WAN Setting
1. IP Type
a. Fixed IP (Default IP: 192.168.0.100)
b. DHCP Client
c. PPPoE
2. Main IP: The current IP address. The IP chaning need to under the
Fixed IP mode.
3. PPPoE Setting
The PPPoE Configuration item is to setup the PPPoE Username and
Password. If you have PPPoE account from the Service Provider, please
input the Username and the Password correctly
After the setting, please click submit and save change button and wait for
system reboot
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11.2 SNTP Settings
User 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.
SNTP settings (Default: On)
After the setting, please click submit and save change button and wait for
system reboot
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12. SIP Setting
User 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 information correctly then you can
register to SIP Proxy Server correctly.
12.1 Service Domain Setting
In Service Domain Function you need to input the account and the
related information in this page please refer to your ISP Provider.
You can register three SIP accounts. You can dial the VoIP phone
to your friends via first enable SIP account and receive the phone
from the tree SIP account.
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(1) Active: On /OFF
(2) Display name: you can input the name you want to display.
(3) User name: you need to input the User Name get from your ISP.
(4) Register Name: you need to input the Register Name get from your
ISP.
(5) Register Password: you need to input the Register Password get
from ISP.
(6) Domain Server: you need to input the Domain Server get from your
ISP.
(7) Proxy Server: you need to input the Proxy Server get from your ISP.
(8) Outbound Proxy: you need to input the Outbound Proxy get from your
ISP. If your ISP does not provide the information, then you can skip
this item.
(9) Status: Register or Not register
After the setting, please click submit and save change button and wait for
system reboot
Example:
Register VoipBuster
Your Voipbuster username
Your Voipbuster password
Proxy Server’s IP
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12.2 Ports Setting
Internal Dial Peer Port: default = 5060 (*important* this port number can’t coincide
with SIP port or RTP port)
SIP port: default = ch1:5064 ch2:5066 ch3:5068…etc (*important* this port number
can’t coincide with dial peer port or RTP port)
You can only change the port number on Ch1; other Channels will be changed
automatically
RTP port: default = ch1:20004 ch2:20006 ch3:20008…etc (*important* this port
number can’t coincide with dial peer port or SIP port)
You can only change the port number on Ch1; other Channels will be changed
automatically
After the setting, please click Submit and save change button to wait for system
reboot
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12.3 Codec Settings:
User can setup the Codec priority, RTP packet length in this page.
Please follow the ISP suggestion to setup these items.
RTP Packet Length
1. G.711& G.729: Default is 20ms.
Range: 10ms,20ms,30ms,40ms,50ms,60ms,70ms,80ms,90ms
2. G.723: Default:
Range: 30ms ,60ms, 90ms
After the setting, please click Submit and save change button to wait for
system reboot
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12.4 Codec ID Setting
User can setup the Codec ID in this page.
After the setting, please click Submit and save change button to wait for
system reboot
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12.5 DTMF Setting
1. Format:
a. 2833: Default RFC2833, the type of DTMF Data Transfer Format
b. Inband: The Type of Inband DMTF Data Transfer Format
c. SIP Info: The Type of SIP-Info DMTF Data Transfer Format;
2. Duration: Default is -1. It’s the duration for MV-3716/MV-3732 to
defect sender’s DTMF. If the parameter is 0, MV-3716/MV-3732 won’t
detect sender’s DTMF. Parameter is 0~999 seconds. After that
duration, MV-3716/MV-3732 won’t detect DTMF.
3. Debounce: Default is 80ms.User can adjust for own. If DTMF is
adding more digits, please increase parameter over 80. If DMTF is
lost digit, please decrease parameter less than 80.
After the setting, please click Submit and save change button to wait for
system reboot
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12.6 SIP Responses
Mobile Busy Response
1. Unavailable: User can setup the SIP response code ofLAN side while
the call dial failed or in busy line
a. 486 Busy Here (Default)
b. 503 Service unavailable
c. 480 Temporarily unavailable
2. Ring Timeout: User can setup the response SIP code of LAN side
while operators hang up the no answered calls
a. 486 Busy Here (Default)
b. 503 Service unavailable
c. 480 Temporarily unavailable
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SIP Ring Response
1. 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
voice directly. (For this function, 183 must be turn on)
2. 183(Session Progress)
[It means "on progressing"]: When you turn 183 on, it means you can
hear the prompt voice while GSM side is busy we recommend you to turn
this on if you use SIP Proxy.
After the setting, please click Submit and save change button to wait for
system reboot
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12.7 Other Settings
User can setup the Hold by RFC and QoS in this page. To change these
settings, please follow your ISP information. The QoS setting is to set the
voice packets’ priority. If you set the value higher than 0, then the voice
packets will get the higher priority to the Internet. But the QoS function
still need to cooperate with the others Internet devices.
1. Hold RFC of Mobile:
a. On: To activate Hold RFC of Moible
b. OFF (Default)
2. Voice QoS : The setting of Voice QoS, Default is 40
3. SIP QoS : The setting of SIP QoS, Default is 40
4. SIP Expire Time : The setting of SIP Expire Time, Default is 40
After the setting, please click Submit and save change button to wait for system
reboot
You can click Submit All to copy to Mobile setting, and select Yes and save change to
wait for the system reboot
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13. STUN Setting
User 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. Please following your ISP information to change the
settings
Public STUN OFF Default is OFF; While the WAN setting of
MV-3716/MV-3732 is in Static IP or Private IP
please selects Public STUN OFF.
Public STUN ON While MV-3716/MV-3732 is working under Firewall
or behind NAT, It will cause SIP can’t register, or
one side communicate, please select Public
STUN ON.
STUN Server The STUN Server IP (Default: stun.iptel.org)
STUN Port The STUN Port (Default: 3478)
After the setting, please click Submit and save change button to wait for
system reboot
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14. Update
14.1 Update Firmware
User can update the system’s firmware to the new one or the factory
reset to let the system back to default setting.
NOTE: Please open the webpage from Internet Explorer, not compatible with FF or Google
Chrome
Step:
(1) Select the firmware code type, Risc code only.
(2)Click the “Browse” button in the right side of the File Location or you
can type the correct path and the filename in File Location bLANk.
(3)Select the correct file you want to download to the system then click
the Update button.
(4) Please click update/default setting after update firmware
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14.2 Default Settings
1. Restore default settings: User can restore the factory default settings
to the system. All setting will restore default setting.
The device IP still is the user original IP.
2. Restore factory all settings: All setting will be restored to default
setting. The device IP will be back to 192.168.0.100
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15. System Authority
User can change the login name and password
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16. Save Change
User can save the changes after the setting is 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
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17. Reboot
User can restart the system. If you want to restart the system, you can
just click the Reboot button, and then the system will automatically.
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18. Specification
18.1 Protocols
SIP (RFC2543, RFC3261)
18.2 TCP/IP
IP/TCP/UDP/RTP/RTCP/
CMP/ARP/RARP/SNTP
DHCP/DNS Client
IEEE802.1P/Q
ToS/DiffServ
NAT Traversal
STUN
uPnP
IP Assignment
Static IP
DHCP
PPPoE
18.3 Codec
G.711 u-Law
G.711 a-Law
G.729A
G.729A/B
18.4 Voice Quality
VAD
CNG
AEC, LEC
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Packet loss
18.5 GSM (MV-3716/MV-3732)
Quad Band: 900/1800/1900/850MHZ
3G/UMTS: for all world and Japan (SoftBank and Docomo)
3G: EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850,
1900, 2100 MHz
CDMA 2000(800MHZ/1900MHZ)
**Please note**
1. Most CDMA -2000 operators don't offer Answer signal.
So VoIP to Mobile, MV-3716/MV-3732 will connect soon.
CDMA -2000 operators will start billing soon. It doesn't wait
mobile side answer
2. CDMA Version doesn't support SMS Feature and 180/183
unavailable
3. CDMA version doesn't have Remote SIM feature
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19. Simple Steps
Step 1. Change the Network setting as you need (Network/network setting)
Step 2. Register SIP proxy Server or Asterisk or VoipBuster as you need
(sip setting/service domain)
Step 3. Set Mobile setting –adjust your gain as you need
Step 4. Set Route ( request )
mobile to LAN:
(1)
*,* --->it is two stage dialing.
when mobile call in,MV-37x 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.
(2)
*, specific extension or IP or phone number
when mobile call in,MV-37x will connect with this specific extension or
IP or phone number auto
*
If you want to set specific phone number, please note your asterisk
need to have route of destination number.
LAN to Mobile:
(1)
*,* --->it is two stage dialing.
When LAN phone call in, MV-37x will provide dial tone and you can
enter mobile number.
(2)
*, specific mobile number
When LAN phone call in, MV-37x will connect with the specific mobile
number auto.
(3)
*,#--->It is 1 stage dialing
When LAN phone and MV-37x both register Asterisk,
you can dial any destination number from LAN phone directly.
*
Please note: Asterisk need to set route of destination number that
dial out from MV-37x
* All changes both need to click "save and change"
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20. Appendix: Setup MV-37x with Asterisk
MV-37x Settings
Can register Asterisk or not
Asterisk want to transfer
CLID,please choose Tel/Tel
(Not Reg)
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Set your Asterisk IP or extension or *
As Asterisk GSM
Route
-87-
-88-
Don't forget to Save changes and then reboot
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Asterisk / Trixbox setting
Add SIP Trunk:
Type your mobile
number
MV-374: 4
MV-378: 8
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Type a trunk
name
Type MV-37X’s IP and port
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Set GSM Route that dial out via MV-37X
Frequency: Quad Band:900/1800/1900/850MHZ
GSM Module use Simcom sim340
Compliant to GSM phase 2/2+
-Class 4 (2W@850/900 MHz)
-Class 1 (1W@1800/1900 MHz)
The prefix of route :
09XXXXXXXX
After change, please press “Sumbit
changes” and “apply configuration
changes”
Choose SIP Trunk
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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. These limits are designed to
provide reasonable protection against harmful interference in a residential
installation. This equipment generates, uses and can radiate radio frequency energy
and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that
interference will not occur in a particular installation. If this equipment does cause
harmful interference to radio or television reception, which can be determined by
turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:
-Reorient or relocate the receiving antenna.
-Increase the separation between the equipment and receiver.
-Connect the equipment into an outlet on a circuit different from that to which the
receiver is connected.
-Consult the dealer or an experienced radio/TV technician for help.
Operation is subject to the following two conditions:
1) this device may not cause interference and
2) this device must accept any interference, including interference that may cause
undesired operation of the device.
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FCC RF Radiation Exposure Statement:
1. This Transmitter must not be co-located or operating in conjunction with any
other antenna or transmitter.
2. 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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