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SunComm SC-375, User Manual

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SunComm SC-375 User Manual Download Page 44

 

 

-42- 

 

 

 

Here the '#' is important to avoid the two stage dialing when you give a 
call from Asterisk to GSM.  
 

Summary of Contents for SC-375

Page 1: ...SunComm SC 375 Nov 2007 Edition User s Manual...

Page 2: ...WEB PAGE 5 7 SYSTEM INFORMATION 5 8 ROUTE 6 9 MOBILE 12 10 NETWORK 17 11 SIP SETTING 20 12 NAT TRANS 30 13 SYSTEM AUTH 31 14 SAVE CHANGE 32 15 UPDATE 33 16 REBOOT 35 17 SETTING AND CHECKING VIA IVR 3...

Page 3: ...ets of SC 375 you may even build an international call network 2 Functions 2 1 VoIP SIP GSM conversion 2 2 VoIP SIP CDMA conversion 2 3 Voice response for setting and status enquiring Dial in GSM numb...

Page 4: ...1 SC 375 main body 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 When you receive SC 375 package and find it is damaged or incorrect please contac...

Page 5: ...system s power supply is normal 4 5 MOBILE GSM indicator light green light Light flashes when GSM status is normal light turns on constantly when GSM is called 4 6 LAN LAN indicator light green light...

Page 6: ...nect the antenna and place it in a good receiving location not too close to the device 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 shou...

Page 7: ...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 System Information 7 1 After login...

Page 8: ...or select the item according to your situation Note Please remember to save change whenever you submit any setting Click Save Change then Save button the system will restart and make the changed funct...

Page 9: ...Please note the priority of the routing rules the CID with more digits gets the priority 8 1 2 URL The IP address of destination You could set the URL as the following formats 1 The complete IP addre...

Page 10: ...server Asterisk has the route of 09 2 Mobile to Lan Any incoming call gets a prompt dial tone so the caller can enter any IP address sip extension or phone number Precondition a SIP extension or phone...

Page 11: ...pt 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...

Page 12: ...he incoming call 8 3 1 URL The IP address or proxy extension numbers of the incoming call You could set the URL as the following formats 1 The complete IP address e g 192 168 0 101 2 The proxy extensi...

Page 13: ...er asterisk has set the routing rules to assign specific prefix of numbers to be transferred from SC 375 3 Lan to Mobile routing sets Usage You could dial on your lan phone call any destination number...

Page 14: ...ard been registered 2 SIM Card ID SIM card ID 3 Signal Quality Signal quality 4 GSM S N IMEI Number 5 Incoming IP The IP address of the last incoming call from LAN 6 Incoming IP Name proxy server name...

Page 15: ...option to fit your need You could leave those default value before you had tried the complete operation of this device 3 LAN Dialtone Gain DTMF Reciver is not good you can adjust gain down 1 2 3 5 6 7...

Page 16: ...he 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...

Page 17: ...Mobile Forward Setting When the first route are busying SIP can transfer phone call to another free route When the device are busying the phone call can be transfer to another device external equipme...

Page 18: ...s 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...

Page 19: ...ings 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 The default IP is...

Page 20: ...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: ...19 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...

Page 22: ...ng Service Domain In this page you should input the data refer to your ISP Maximum is 3 accounts Realm 1 to 3 You could dial out via first SIP account and receive via the three SIP accounts 1 Active c...

Page 23: ...put 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 Remember to click Save Charge 10 You can see the Registe...

Page 24: ...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 c...

Page 25: ...23 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...

Page 26: ...24 11 4 Codec ID Setting You can setup the Codec ID in this page...

Page 27: ...25 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...

Page 28: ...26 11 6 RPort Setting You can setup the RPort Enable Disable according to your ISP information After setting remember to click the Submit button...

Page 29: ...n 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 cal...

Page 30: ...se 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 The QoS setting is to set the voice packets priorit...

Page 31: ...29...

Page 32: ...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 sett...

Page 33: ...31 13 System Auth In this page System Authority you could change your login name and password...

Page 34: ...32 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...

Page 35: ...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 Ple...

Page 36: ...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 settin...

Page 37: ...35 16 Reboot In this page you could click the Reboot button to restart the system...

Page 38: ...100 4 Check IP Type 121 IVR announces DHCP is on or off Default off 5 Check Network Mask 123 IVR announces the current network mask Default 255 255 255 0 6 Check Gateway IP Address 124 IVR announces...

Page 39: ...Address 114xxx xx x xxx xxx Must set Static IP first Enter IP address using numbers on the telephone keypad Use the star key when entering a decimal point 13 Set Primary DNS Server 115xxx xx x xxx xx...

Page 40: ...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 723 1 5 3k G 723 1 6 3k G 729A G 729A B 18 4 Voice Quality VAD CNG AEC...

Page 41: ...1 SC 375 mobile to lan set route table Step 2 no 2 SC 375 lan to mobile set route table Step 3 Additionally two pcs SC 375 both need to register proxy server Step 4 And proxy server set the route tha...

Page 42: ...er when mobile call in SC 375 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...

Page 43: ...h your mobile operator like free special number you can call your SC 375 for free You can then call all around the world from your mobile at voip cost 21 2 SC 375 Configuration Once you ve configured...

Page 44: ...42 Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM...

Page 45: ...r 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...

Page 46: ...44 Once Asterisk configuration is made you should get Registered on the Realm1...

Page 47: ...ed 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...

Page 48: ...to NO if behind NAT insecure very qualify yes disallow all allow ulaw prefered codec for DTMF detection allow alaw 21 6 extensions conf GSM Gateway incoming calls gateway exten _103 1 Answer exten _10...

Page 49: ...60 289a z X Lite 1105x Modify file z 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...

Page 50: ...en 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...

Page 51: ...2 168 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 Contact sip 1002 192 16...

Page 52: ...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 tag 677373503 Contact sip 1001 192 168 66 145 733...

Page 53: ...h z9hG4bK4C4315351FC84CA582D14FB8C25F C3BF From user_1001 sip 1001 192 168 66 202 7331 tag 1121869743 To sip 1002 192 168 66 202 Contact sip 1001 192 168 66 145 7331 Call ID F4B32CA6 1835 4E68 941A C6...

Page 54: ...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 Al...

Page 55: ...e36d8f1 To sip 1002 192 168 66 202 Call ID 7e45b773130f1fc945efcee502f84042 192 168 66 203 Contact sip 1002 192 168 66 203 5060 CSeq 10 REGISTER Expires 300 Authorization Digest username 1002 realm as...

Page 56: ...8 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 7e45b773130...

Page 57: ...6 202 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 SI...

Page 58: ...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 Via SIP 2 0 UDP 192 168 66 203 5060 branch z...

Page 59: ...57 SunComm Technology Co Ltd Tel 886 2 32341496 Fax 886 2 32341393 E mail sales2 suncomm com tw Website www suncomm com tw...

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