SunComm SC-385 User Manual
SunComm SC-385 is a 2-channel VoIP GSM gateway that can make and receive calls between landlines and mobile networks. It supports SIP protocol and is compatible with Asterisk. With 50 configurable routes for mobile-to-landline and landline-to-mobile calls, it offers flexible call forwarding options. It features voice response for status and settings, allowing users to manage the device remotely. The device also supports series connections for cost savings.
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SC-385
GSM VoIP Gateway
2 channels
User Manual
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Content
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1.INTRODUCTION ................................................................................................................. 1
2.FUNCTION DESCRIPTION............................................................................................... 1
3.PARTS LIST.......................................................................................................................... 1
4.DIMENSION ......................................................................................................................... 2
5.CHART OF THE DEVICE .................................................................................................. 3
6.CABLING .............................................................................................................................. 4
7. IP SETTING ......................................................................................................................... 5
8.WEB PAGE SETTING......................................................................................................... 7
9.SYSTEM INFORMATION.................................................................................................. 8
10. ROUTE................................................................................................................................ 9
11.MOBILE ............................................................................................................................ 14
12.NETWORK........................................................................................................................ 17
13.SIP SETTING .................................................................................................................... 21
14. NAT TRANS ..................................................................................................................... 29
15.SYSTEM AUTH. ............................................................................................................... 30
16.SAVE CHANGE................................................................................................................ 31
17.UPDATE ............................................................................................................................ 32
18.REBOOT............................................................................................................................ 34
19.SPECIFICATION ............................................................................................................. 35
20.SETUP SC-385 WITH ASTERISK ................................................................................. 36
1. Introduction
SC-385 is a 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 make 2 calls simultaneously from SIP VoIP devices to GSM networks and GSM network to VoIP devices.
2. Function description
2.1 VoIP(SIP) 、 GSM(SC-385) 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 ，
Communicates with other gateway or Softphone via PC or notebook
3. Parts list
Please check the parts for any missing parts. If do, please contact our agents ：
3.1 「 SC-385 」 main body
3.2 Power adaptor AC-DC (110V AC – 12V DC) or (220V AC – 12V DC)
3.3 Network cable
3.4 Antenna
3.5 User Manual
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4. Dimension
17cm
(1)
(3)
14.5cm
(2)
(4)
4.1cm
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5. Chart of the device
5.1
5.2 5.3 5.4 5.5 5.6 5.7 5.8
5.1 Antenna ： Antenna connector
5.2 DC 12V ： Power input.
5.3 LAN ： LAN port: It also can be DHCP Server.
5.4 WAN: RJ-45 internet connector ， standard RJ-45 socket ， connect to
HUB.
5.5 PWR (Power LED) ： Light up when power is normal.
5.5 VoIP1 ： An indicator light of VoIP1
5.6 VoIP2 ： An indicator light of VoIP2
5.8 LINK Indicator ： Light up when network is connected.
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6. CABLING
6.1 Connect the internet cable from HUB to the ‘WAN’ connector of the
SC-385.
*If you need to stack up more SC-385,you can stack up as follows.
6.2 Connect the antenna and put it in proper position to get the best signal reception.
6.3 Insert the SIM card from back of the main body. (take the slide off first).
6.4 Connect the power adaptor. The ‘POWER’ LED should be light up.
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7. 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. The operator may dial in the mobile number during this period to set or query the network parameters.
Item
5 Check Network
Mask
IP
IVR Action IVR Menu Choice
1 Reboot
3 Check
4 Check
#195#
#120#
Type #121#
#123#
Notes
After you hear “Option
Successful,” hang-up. Unit will reboot automatically.
System will automatically
Reboot. WARNING: ALL
User-Changeable”
NONDEFAULT SETTINGS
WILL BE LOST! This will include network and service provider data.
IVR will announce the current
IP address , Default :
192.168.0.100
IVR will announce if DHCP is enabled or disabled. default : OFF
IVR will announce the current network mask. Default :
255.255.255.0
IVR will announce the current gateway IP address,
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7
8
10
Check Primary
DNS Server
Check Firmware
Version client
Set Static IP
Address
#125#
#128#
#112xxx*xxx*xxx
*xxx#
11 Set Network Mask #113xxx*xxx*xxx
*xxx#
12
13
Set Gateway IP
Address
Set Primary DNS
Server
#114xxx*xxx*xxx
*xxx#
#115xxx*xxx*xxx
*xxx#
IVR will announce the current setting in the Primary DNS field.
Default : 192.168.0.1
IVR will announce the version of the firmware running
The system will change to
DHCP
Client type
DHCP will be disabled and system will change to the
Static IP type.
Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.
Must set Static IP first.
Enter value using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.
Must set Static IP first.
Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.
Must set Static IP first.
Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.
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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) . The page follows when the username and password are correct.
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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.
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10. Route
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 Ssc-385 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. 0911* means any number starting with 0911 will be accepted
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(3) * means all numbers can be accepted
(4) N means the calls without the CID
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.
*URL ： The IP address to transfer this call
(1) may enter the whole IP address, e.g. 192.168.0.101 or proxy extension .
(2) If this field is blank or simply ‘N’, it means refuse to transfer.
(3) If an ‘*’ entered, it means 2-stages-dialing. The call will be answered and prompt dial tone again to receive the IP address as the destination. The caller may enter the IP such as 192*168*0*101#.
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10.2 Mobile to LAN Speed Dial Settings
When you set Mobile to LAN Speed Dial Settings and Mobile to
LAN at the same time,SC-385 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” as destination.
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10.3 LAN to Mobile Settings
The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE.
The SC-385 will transfer to the mobile number according to the incoming
URL
*URL ： The IP address of the incoming call 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.
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*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. #['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 #d2a09 means one-stage dialing, delete the first 2 codes from your destination number, then add 09 in front as the new destination number.
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11. Mobile
11.1 Mobile Status
(1)Network Registration ： The telecom carrier which the SIM card has been registered
(2)SIM Card ID ： SIM card ID.
(3)Signal Quality ： Signal quality.
(4)Incoming IP ： The IP address of the last incoming call from LAN
(5)Incoming IP Name: proxy server name
(6)Outgoing IP ： The IP address of the last outgoing call to LAN
(7)Incoming Mob ： The caller ID of the last incoming call from MOBILE
(8)Outgoing Mob ： The called number of the last outgoing call to MOBILE
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11.2 Mobile Setting
(1)
(3)
(4)
(6)
(7)
(8)
(9)
(10)
(2)
(5)
LAN
VoIP
Mobile 1:
(5)Rx
(4) Tx
Codec
(1)VoIP Tx Gain
(2) VoIP Rx Gain
Mobile 2:
Rx
Codec
Tx
DTMF
DTMF
GSM
GSM
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(3)LAN Dial tone Gain: DTMF Receiver is not good, you can adjust gain down.
(4)CODEC Tx Gain: as above
(5)CODEC Rx Gain: as above
(6)Caller ID: You may select to display the Caller ID from GSM incoming call, or fixed SIP user name.
(7)Presentation CLIR: If you need to block the Caller Id for call termination, please choose Suppression
(8)Mobile PIN Code: If you need to unlock pin code via SC-385, you can click “On” and enter pin code.
(9)LAN Answer Mode:
Answered: when mobile answer, then connect the call
Alerted: when the mobile is ringing back tone, then connect the call
Income: when LAN dial out, then connect soon
(10)Band Type: When you buy Quad band, you need to choose your
GSM frequency
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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.
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12.2 WAN 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) The PPPoE Configuration item is to setup the PPPoE Username and
Password. If you have the PPPoE account from your Service
Provider, please input the Username and the Password correctly.
(3) The Bridge Item is to setup the system Bridge mode Enable/Disable.
If you set the Bridge On, then the two Fast Ethernet ports will be transparent.
(4) When you finished the setting, please click the Submit button.
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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
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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. When you finished the setting, please click the Submit button.
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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 information correctly then you can register to SIP Proxy Server correctly.
13.1 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.
First you need to click Active to enable the Service Domain, then you can input the following items.
(1)No.,: choose Mobile 1 or Mobile 2
(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) You can see the Register Status in the Status item.
(10) When you finished the setting, please click the Submit button.
Remember to click “Save Charge”
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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.
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13.4 Codec ID Setting
You can setup the Codec ID in this page.
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13.5 DTMF Setting
You can setup the DTMF Setting in this page.
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13.6 RPort Function:
You can setup the RPort Enable/Disable in this page. To change this setting, please follow your ISP information. When you finished the setting, please click the Submit button.
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13.7 Other Settings
Other Settings: you can setup the Hold by RFC and QoS in this page. To change these setting, please follow your ISP information. When you finished the setting, please click the Submit button. 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.
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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. To change these settings please following your ISP information. When you finished the setting, please click the Submit button.
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15. System Auth.
In System Authority you can change your login name and password.
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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.
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17. Update
In Update you can update the system’s firmware to the new one or you can 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. You can upgrade the firmware by the following steps:
(2)Select the firmware code type, Risc code.
(3)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.
(4)Select the correct file you want to download to the system then click the Update button.
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17.2 Restore Default Settings
Default Setting, you can restore the system to factory default in this page.
You can just click the Restore button, then the system will restore to default and automatically restart again.
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18. Reboot
Reboot function you can restart the system. If you want to restart the system, you can just click the Reboot button, then the system will automatically.
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19. Specification
19.1 Protocols
SIP (RFC2543, RFC3261)
19.2 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
19.3 Codec
G.711 u-Law
G.711 a-Law
G.723.1 (5.3k)
G.723.1 (6.3k)
G.729A
G.729A/B
19.4 Voice Quality
VAD
CNG
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AEC, LEC
Packet loss
19.5 GSM (SC-385)
Dual BAND: 900/1800 MHZ
Tri BAND: 900/1800/1900 MHZ
Quad BAND: 900/1800/1900/850 MHZ
20.Setup SC-385 with Asterisk
20.1 Usage
A typical usage of such a gateway is to be able to give a call with your normal mobile to any destination at voip cost:
Your mobile <----GSM network ----> SC-385 <--LAN--> Asterisk
<-internet --> VOIP provider <-whatever --> landline
To do such a call, you just call your SC-385 number (it has its own SIM
CARD), then you get an invitation tone, then you dial the number which is handled by Asterisk.
If you have some special deals with your mobile operator, like free special number, you can call your SC-385 for free.
You can then call all around the world from your mobile at voip cost :-)
20.2 SC-385 Configuration
Once you've configured everything in the box, one good advice is to unplug the power and to restart it. By this way you should have all the parameters taken into account.
To have the SC-385 to work with Asterisk, you need first to configure the box.
Here are some screen shots showing all the important parameters.
You have to note that in all the configuration process, the SC-385 is
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considered as extension '103' of the IPBX.
In Bold are the parameters depending on your installation
Here the '#' is important to avoid the two stage dialing when you give a call from Asterisk to GSM.
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The mobile numbers you give in that page are the authorized mobile which can call GSM to Asterisk.
These mobile numbers must be defined as your GSM provider displays the number.
If you don't know how it is displayed, just give a call to the box and check the number given in the 'Incoming Mob' field of the 'Mobile Status' page.
Any number which is not in that list won't have access to the LAN side, so to Asterisk.
If you want to allow any number, just set '*' in that field ... but beware of the bill ;-)
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Once Asterisk configuration is made, you should get 'Registered' on the
Realm1.
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It is very important to use only ulaw or alaw as all DTMF is inband.
So if you want to be able to do some DISA when you call from GSM to
Asterisk, it has to be one of these 2 codecs.
These settings seem to be ok, just adjust ...
20.3 Antenna position
Another important thing is to properly place the provided antenna.
If your GSM reception is good, you should get around 18 or 19 as Signal
Quality in the "Mobile Status" page.
With that level of signal quality, your audio quality will be very good.
On the other end, I've experienced that with a signal quality down to 11, audio becomes very jerky.
So, maximum signal quality = maximum audio quality.
20.4 Asterisk configuration
Once the SC-385 is set, you have to configure Asterisk.
On that side, you have to setup files as follow :
20.5 sip.conf
; GSM VOIP Gateway SC-385
[103]
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type=friend username=103 fromuser=103 regexten=103 ; When they register, create extension 401 secret=xxxxxxx ; Asterisk extension password context=gateway ; Incoming calls context dtmfmode=inband ; Very important for DISA to work call-limit=1 ; Limit to 1 call max callerid=GSM Gateway <103> host=dynamic nat=no ; Gateway is not behind a NAT router canreinvite=no ; Typically set to NO if behind NAT insecure=very qualify=yes disallow=all allow=ulaw ; prefered codec for DTMF detection allow=alaw
20.6 extensions.conf
; ******* GSM Gateway incoming calls **********
[gateway] exten => _103,1,Answer() exten => _103,2,DigitTimeout(3) ; give enough time to do second stage dialing exten => _103,3,ResponseTimeout(5) exten => _103,4,DISA(no-password|outgoing) ; here 'outgoing' is the normal context to deal with the dial plan
[outgoing]
...
; example of LAN to GSM call
; call the SC-385 sim card mail box thru GSM exten => _888,1,SetCallerID("xxxxxxxxxx") exten => _888,2,Dial(SIP/${EXTEN}@103,60,r) exten => _888,3,Hangup()
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