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SunComm SC-385 User Manual

Gsm voip gateway 2 channels
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SC-385
GSM VoIP Gateway
2 channels
User Manual

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Table of Contents
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Summary of Contents for SunComm SC-385

  • Page 1 SC-385 GSM VoIP Gateway 2 channels User Manual...

  • Page 2: Table Of Contents

    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...

  • Page 3: Introduction

    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.

  • Page 4: Dimension

    4. Dimension 4.1cm 17cm 14.5cm...

  • Page 5: Chart Of The Device

    5. Chart of the device 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.

  • Page 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.

  • Page 7: Ip Setting

    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.
  • Page 8 Address Default : 192.168.0.254 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...

  • Page 9: Web

    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).

  • Page 10: System Information

    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.

  • Page 11: Route

    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：...
  • Page 12 (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.
  • Page 13 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”...
  • Page 14 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：The IP address of the incoming call may enter the whole IP address, e.g. 192.168.0.101 or proxy server’s extension.
  • Page 15 *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.

  • Page 16: Mobile

    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...
  • Page 17 11.2 Mobile Setting (10) Mobile 1: (5)Rx VoIP Codec (4) Tx DTMF Mobile 2: (1)VoIP Tx Gain Codec (2) VoIP Rx Gain DTMF -15-...
  • Page 18 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:...

  • Page 19: Network

    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. -17-...
  • Page 20 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.
  • Page 21 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 -19-...
  • Page 22 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.

  • Page 23: Sip Setting

    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.
  • Page 24 -22-...
  • Page 25 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.
  • Page 26 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. -24-...
  • Page 27 13.4 Codec ID Setting You can setup the Codec ID in this page. -25-...
  • Page 28 13.5 DTMF Setting You can setup the DTMF Setting in this page. -26-...
  • Page 29 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. -27-...
  • Page 30 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’...

  • Page 31: Nat Trans

    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.

  • Page 32: System Auth

    15. System Auth. In System Authority you can change your login name and password. -30-...

  • Page 33: Save Change

    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.

  • Page 34: Update

    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.
  • Page 35 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. -33-...

  • Page 36: Reboot

    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. -34-...

  • Page 37: Specification

    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 -35-...

  • Page 38: Setup Sc-385 With Asterisk

    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.
  • Page 39 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. -37-...
  • Page 40 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.
  • Page 41 Once Asterisk configuration is made, you should get 'Registered' on the Realm1. -39-...
  • Page 42 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...
  • Page 43 => _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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