SunComm SC-375 User Manual
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SunComm SC-375 User Manual

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SunComm
SC-375
(Nov., 2007 Edition)
User's Manual

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Summary of Contents for SunComm SC-375

  • Page 1 SunComm SC-375 (Nov., 2007 Edition) User’s Manual...

  • Page 2: Table Of Contents

    15.UPDATE ..........................33 16.REBOOT..........................35 17. SETTING AND CHECKING VIA IVR................. 36 18.SPECIFICATION ......................38 19. APPLICATIONS......................39 20. SIMPLE STEPS ....................... 40 21. APPENDIX: SETUP SC-375 WITH ASTERISK ............41 22.HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM ......47...

  • Page 3: Introduction

    PSTN） You may use a SIP-protocol VoIP phone or software to connect to the SC-375, then reach this call to the mobile network, and vice versa. With multiple sets of SC-375, you may even build an international call network.

  • Page 4: The Contents In Package

    3.2 AC-DC Adaptor (110V AC – 12V DC or 220V AC – 12V DC) 3.3 Network cable 3.4 Antenna 3.5 User’s Manual (3.1) (3.2) (3.4) (3.3) When you receive SC-375 package and find it is damaged or incorrect, please contact your vendor.

  • Page 5: Dimension And Panel Description

    4. Dimension and Panel description 17cm 4.1cm 14.5cm (4.1) (4.2) (4.3) (4.4) (4.6) (4.5) (4.7) 4.1 Antenna：Antenna connector. 4.2 DC 12V：Power socket. 4.3 LAN: Standard RJ-45 socket, connecting to Hub circuit. 4.4 PWR: Power indicator light, red light. Light is on when system’s power supply is normal.

  • Page 6: Accessory Attachment

    5.3 Insert a SIM card into back of SC-375. 5.4 Plug the adapter in DC 12V socket and PWR socket. The PWR light should turn red at the moment. 5.5 Click reset button 3 sec. SC-375 will restore default IP. Other setting as usual.

  • Page 7: Setting And Managing Via Web Page

    6. Setting and managing via web page The default IP address of SC-375 is http://192.168.0.100. Before accessing the web page, please confirm this address is available in your network. Enter the default username and password to login. Default username: voip Default password: 1234 7.

  • Page 8: Route

    7.2 You could also see the setting table in the left side. Please click on the option you would like to set. The setting methods are indicated as the following chapters, please input the value or select the item according to your situation. Note: Please remember to save change whenever you submit any setting.
  • Page 9 SC-375 When the GSM number of the is called, this device transfers the call to URL according to the caller ID of the incoming call. 8.1.1 CID: caller ID, the numbers of incoming call You could set the CID as the following formats: (1) The complete number, e.g.
  • Page 10 8.1.3 Example of Mobile to Lan setting: (1) Mobile to Lan: 0932*, 0911123456 When the GSM numbers of the device is called, if the caller’s prefix SC-375 numbers are 0932, transfers the call to 0911123456, then 0911123456 rings (while available).
  • Page 11 The call is answered with a prompt dial tone for the caller to press the “Num”, and then the device connects the “URL” as destination. Example: after you call the GSM number of the device and hear a dial tone, you press 0, then the lan phone of IP address: 192.168.0.107 rings.
  • Page 12 8.3 Route/ LAN to Mobile Settings In this page: Lan To Mobile able, you could set the routing rules to transfer the calls incoming from Lan to Mobile. Maximum 50 sets. SC-375 When the Lan of the is called, this device transfers the call to Call Num according to the URL of the incoming call.
  • Page 13 0911111111 , this allow the caller with lan phone dial directly the destination numbers. Precondition: SC-375 and incoming lan Phone are both registered at proxy server or Asterisk. (2) Proxy server/asterisk has set the routing rules to assign...

  • Page 14: Mobile

    9. Mobile 9.1 Mobile/ Mobile Status In this page: Mobile Status, you could get the information of your GSM network and the latest operation. (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.
  • Page 15 9.2 Mobile/ Mobile Setting In this page: Mobile setting, you could adjust the parameter and click on the option to fit your need. You could leave those default value before you had tried the complete operation of this device. (10) (11) (6)Rx VoIP...
  • Page 16 Please only fill proxy server ip,Username and choose Active: on (else field empty) in sip setting/service demain (8)Presentation CLIR : If you need to block the Caller Id for call termination,please choose Suppression (9)Mobile PIN Code:If you need to unlock pin code via SC-375,you can -14-...
  • Page 17 click “On” and enter pin code. (10)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 (11)Band Type:When you buy Quad band,you need to choose your GSM frequency 9.3 Mobile / Forward Setting : When the first route are busying, SIP can transfer phone call to...
  • Page 18 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).

  • Page 19: Network

    10. Network In Network, you could check the Network status; configure the WLAN Settings, LAN Settings and SNTP settings. 10.1 Network/ Status/ Network Status: information of current Network in this page. 10.2 Network/ Network Settings/ Lan Settings: You can check the current Network setting in this page.
  • Page 20 (1) LAN Mode: select NAT (2) Fixed IP: 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. (3) DHCP client: you could refer to your current network environment to configure the system properly (4) PPPoE: If you have the PPPoE account from your Service Provider, please input the Username and the Password correctly.
  • Page 21 date/time information. Also you could set the Time Zone according to your location; and set the time to synchronize. After setting, remember to click the Submit button. -19-...

  • Page 22: Sip Setting

    11. SIP Setting If you need, you could setup the Service Domain, Port Settings, Codec Settings, RTP setting, RPort Setting and Other Settings. If ISP provides the VoIP service, you need to input the related information correctly to register at SIP Proxy Server. 11.1 SIP Setting/ Service Domain: In this page, you should input the data refer to your ISP.
  • Page 23 (6) Domain Server: input the Domain Server IP address. (7) Proxy Server: input the Proxy Server IP address. (8) Outbound Proxy: input the Outbound Proxy IP address. If your ISP does not provide the information, you could skip this item. (9) After setting, click the Submit button.
  • Page 24 11.2 Port Setting You can setup the SIP and RTP port number in this page. Each ISP provider will have different SIP/RTP port setting, please refer to the ISP to setup the port number correctly. After setting, remember to click the Submit button.
  • Page 25 11.3 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. After setting, remember to click the Submit button. -23-...
  • Page 26 11.4 Codec ID Setting You can setup the Codec ID in this page. -24-...
  • Page 27 11.5 DTMF Setting You can setup the DTMF Setting in this page. Note: If this device has registered at SIP Proxy Server/Asterisk, please select “2833”. If not, please select “Inband DTMF”. -25-...
  • Page 28 11.6 RPort Setting: You can setup the RPort Enable/Disable according to your ISP information. After setting, remember to click the Submit button. -26-...
  • Page 29 11.7 SIP Setting: SIP Responses 11.7.1 486(busy here), 503(Service unavailable): When Device are busying, you can select 486 or 505 to response to SIP. 11.7.2 180 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 Voice-Mail directly.
  • Page 30 11.7.3 183(Session Progress)-->[It means"on progressing"] : When you turn 183 on, it means you can hear voicemail while GMS side are busying. We recommend you to turn this on if you use SIP Proxy. 11.8 Other Settings You could setup the RFC and QoS according to your ISP information. After setting, remember to click the Submit button.
  • Page 31 -29-...

  • Page 32: Nat Trans

    12. NAT Trans In this page: NAT Trans./ STUN, you could setup the STUN Enable/Disable and STUN Server IP address. This function helps your VoIP device work properly behind NAT. Change these settings according to your ISP information. After setting, remember to click the Submit button.

  • Page 33: System Auth

    13.System Auth. In this page: System Authority, you could change your login name and password. -31-...

  • Page 34: Save Change

    14.Save Change Please remember this step whenever you submit any setting. Click “Save Change” then “Save” button, the system will restart and make the changed function/setting operative. -32-...

  • Page 35: Update

    15.Update Here you could update the latest firmware and restore the default settings. 15.1 Update/ New Firmware/ Update Firmware Download the latest firmware, then (1) Method: select “HTTP” (2) Code Type: select “Risc”. (3) File Location: Click the “Browse” button in the right side of the File Location for the file.
  • Page 36 15.2 Restore Default Settings In this page: Update/ Default Settings, you could restore the factory default settings to the system. Click the Restore button, then the system returns to default IP http://192.168.0.100 (the other settings e.g SIP setting, mac address remains), and automatically restart. -34-...

  • Page 37: Reboot

    16.Reboot In this page, you could click the Reboot button to restart the system. -35-...

  • Page 38: Setting And Checking Via Ivr

    17. Setting and checking via IVR User could get or set some parameters of the system by dialing in the The status or result is reported via mobile numbers of the device. voice response system. In the first 20 seconds after power-on (when only Mobile light flash), you could dial its mobile numbers.
  • Page 39 x*xxx*xxx# Enter value using numbers on the telephone keypad. Use the * (star) key when entering a decimal point. Set Gateway IP #114xxx*xx Must set Static IP first. Address x*xxx*xxx# Enter IP address using numbers on the telephone keypad. Use the * (star) key when entering a decimal point.

  • Page 40: Specification

    Static IP DHCP PPPoE 18.3 Codec G.711 u-Law G.711 a-Law G.723.1 (5.3k) G.723.1 (6.3k) G.729A G.729A/B 18.4 Voice Quality AEC, LEC Packet loss SC-375 18.5 GSM ( Dual BAND: 900/1800 MHZ Tri BAND: 900/1800/1900 MHZ Quad BAND: 900/1800/1900/850 MHZ -38-...

  • Page 41: Applications

    1.Connect to VoipBuster a). Register VoipBuster account at Service Domain. b). Route setting: Mobile to Lan set: *,* SC-375 When you call in GSM number of , you can enter destination number that will dial out from VoipBuster. (Landline is free, GSM rate...

  • Page 42: Simple Steps

    Mobile to Lan: (1) *,* --->it is two stage dialing. when mobile call in, SC-375 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.

  • Page 43: Appendix: Setup Sc-375 With Asterisk

    Your mobile <----gsm network----> SC-375 <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> landline To do such a call, you just call your SC-375 number (it has its own simcard), then you get an invitation tone, then you dial the number which is handled by Asterisk.
  • Page 44 Here the '#' is important to avoid the two stage dialing when you give a call from Asterisk to GSM. -42-...
  • Page 45 The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk. These mobile number 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 46 Once Asterisk configuration is made, you should get 'Registered' on the Realm1. -44-...
  • Page 47 So, maximum signal quality = maximum audio quality. 21.4 Asterisk configuration Once the SC-375 is set, you have to configure Asterisk. On that side, you have to setup files as follow : 21.5 sip.conf ; GSM VOIP Gateway SC-375...
  • Page 48 => _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-375 sim card mail box thru GSM exten => _888,1,SetCallerID("xxxxxxxxxx") exten => _888,2,Dial(SIP/${EXTEN}@103,60,r) exten => _888,3,Hangup()

  • Page 49: How To Setup Asterisk To Receive Caller Id From

    22.How to setup Asterisk to receive Caller ID from Test version trixbox-2.2 SIP Softphone SJPhone 1.60.289a X-Lite 1105x Modify file Add the following setting to/etc/asterisk/sip.conf [1000] type=friend secret=1000 qualify=yes nat=yes host=dynamic canreinvite=no context=internal [1001] type=friend secret=1001 qualify=yes nat=yes host=dynamic canreinvite=no context=internal [1002] type=friend...
  • Page 50 Add the following setting to /etc/asterisk/extensions.conf [internal] exten => 1000,1,Dial(SIP/1000) exten => 1001,1,Dial(SIP/1001) exten => 1002,1,Dial(SIP/1002) configure: trixbox-2.2: address=192.168.66.202:5060 SJPhone: address=192.168.66.145:5060; username=1000, displayname=user_1000 X-Lite: address=192.168.66.145:7331; username=1001, displayname=user_1001 SC-375: address=192.168.66.203:5060; username=1002, displayname=user_1002 -48-...
  • Page 51 0928492911(mobile number) SC-375 hear the second dial tone,call SoftPhone’s number SoftPhone show pstn caller id This Is X-Lite receiving packet, red word is pstn number. Test ok. INVITE sip:1001@192.168.66.145:7331 SIP/2.0 Via: SIP/2.0/UDP 192.168.66.202:5060;branch=z9hG4bK3d0bbaf7;rport From: "035678238" <sip:1002@192.168.66.202>;tag=as580472a7 To: <sip:1001@192.168.66.145:7331>...
  • Page 52 Content-Type: application/sdp Content-Length: 242 o=root 2737 2737 IN IP4 192.168.66.202 s=session c=IN IP4 192.168.66.202 t=0 0 m=audio 15852 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.66.202:5060;branch=z9hG4bK3d0bbaf7;rport From: "035678238"...
  • Page 53 0-15 a=sendrecv test 2 SoftPhone call 1002 SC-375 hear second dial tone and call pstn pstn answer show caller id-mobile number 0928492911 This Is X-Lite receiving packet. Test ok. INVITE sip:1002@192.168.66.202 SIP/2.0 Via: SIP/2.0/UDP 192.168.66.145:7331;rport;branch=z9hG4bK4C4315351FC84CA582D14FB8C25F...
  • Page 54 c=IN IP4 192.168.66.145 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.66.145:7331;branch=z9hG4bK4C4315351FC84CA582D14FB8C25FC3BF ;received=192.168.66.145;rport=7331 From: user_1001 <sip:1001@192.168.66.202:7331>;tag=1121869743 To: <sip:1002@192.168.66.202>;tag=as2a2fbf98 Call-ID: F4B32CA6-1835-4E68-941A-C685B39C43FF@192.168.66.145 CSeq: 63148 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1002@192.168.66.202>...
  • Page 55 a=fmtp:101 0-16 a=silenceSupp:off - - - - register issue The packet date from Asterisk as follows. Please note, user_1002’s display name don’t appear So the website’s Display Name is not available <-- SIP read from 192.168.66.203:5060: REGISTER sip:192.168.66.202 SIP/2.0 Via: SIP/2.0/UDP 192.168.66.203:5060;rport;branch=z9hG4bK590e92b551233a10a0ae71944c19b5 From: <sip:1002@192.168.66.202>;tag=4e36d8f1 To: <sip:1002@192.168.66.202>...
  • Page 56 eived=192.168.66.203;rport=5060 From: <sip:1002@192.168.66.202>;tag=4e36d8f1 To: <sip:1002@192.168.66.202> Call-ID: 7e45b773130f1fc945efcee502f84042@192.168.66.203 CSeq: 10 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1002@192.168.66.202> Content-Length: 0 Transmitting (NAT) to 192.168.66.203:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.66.203:5060;branch=z9hG4bK590e92b551233a10a0ae71944c19b5aa;rec eived=192.168.66.203;rport=5060 From: <sip:1002@192.168.66.202>;tag=4e36d8f1 To: <sip:1002@192.168.66.202>;tag=as13a32ae8 Call-ID: 7e45b773130f1fc945efcee502f84042@192.168.66.203 CSeq: 10 REGISTER User-Agent: Asterisk PBX...
  • Page 57 Via: SIP/2.0/UDP 192.168.66.203:5060;rport;branch=z9hG4bK672fa67f59c2223275f5ee286d27597a From: <sip:1002@192.168.66.202>;tag=4e36d8f1 To: <sip:1002@192.168.66.202> Call-ID: 7e45b773130f1fc945efcee502f84042@192.168.66.203 Contact: <sip:1002@192.168.66.203:5060> CSeq: 11 REGISTER Expires: 300 Authorization: Digest username="1002",realm="asterisk",nonce="5def9231",response="046a412f4e7ed4 e98fd507416994a80a",uri="sip:192.168.66.202",algorithm=MD5 User-Agent: CMI CM5K Content-Length: 0 --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.66.203 : 5060 (NAT) Transmitting (NAT) to 192.168.66.203:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP...
  • Page 58 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.66.203:5060: OPTIONS sip:1002@192.168.66.203:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.66.202:5060;branch=z9hG4bK7b92dd8a;rport From: "Unknown" <sip:Unknown@192.168.66.202>;tag=as5dee3942 To: <sip:1002@192.168.66.203:5060> Contact: <sip:Unknown@192.168.66.202> Call-ID: 5ebc2211278e2cb7699911ad39454d4e@192.168.66.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 May 2007 03:11:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Transmitting (NAT) to 192.168.66.203:5060: SIP/2.0 200 OK...
  • Page 59 SunComm Technology Co., Ltd. Tel: 886-2-32341496 Fax: 886-2-32341393 E-mail: sales2@suncomm.com.tw Website: www.suncomm.com.tw -57-...

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