DuMV@PCI 2 ports GSM/VoIP PCI Card User Manual PORTech
Contents
1. 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 PORTech od ct Update Firmware You could update the newest firmware PCB mark 2K123B Route mE Mobile Method O HTTP O TFTP Network SIP Settings NAT Transform Code Type Risc v Update File Location Default Settings B l l TFTP Sewer 192 168 1 250 System Authority Save Change U pdate Update Reset 36 16 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 PORTech Restore Default Settings M CTI tral You could click the restore button to restore the factory settings Route Mobile Restore default settings Network SIP Settings NAT Transform Update New Firmware Default Settings System Authority Save Change Reboot 37 17 Reboot Reboot function you can restart the system If you want to restart the system you can just click the Reboor button then the system will automatically PO RTech Reboot System E CTI minae You could press the reboot button to restart the system Route Mobile Reboot system Network SIP Sett2. 168 66 202 gt tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY WWVW Authenticate Digest algorithm MD5 realm asterisk nonce 5def9231 Content Length 0 Scheduling destruction of call 7e45b773130f1fc945efcee502f84042 192 168 66 203 in 15000 ms asterisk1 CLI gt lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 53 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch2 z9hG4bK672fa67159c2223275f5ee286d27597a From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 11 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 5def9231 response 046a412f14e7ed4 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 192 168 66 203 5060 branch z9hG4bK672fa67f59c2223275f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 2
3. 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 e f both 5060 port and 5062 port are busying at same time you can set up Fwd to External then you can transfer the phone call to another designate device E ya 10 4 Mobile SMS Agent PORTech SMS Agent Your CTI Partner Read received SMS Route Port Status Mobile 1 Standby Mobile Mobile 2 Standby Status SEE I SMS Sender Network Dest Num Maximum Number of UCS2 chars for this text box is 70 SIP Settings NAT Transform Message Update System Authority You have 70 UCS2 chars remaining for your description Save Change Reboot 1 Rx List Read received SMS 2 Dest Num the Receiver s phone number 3 Message Please fill the message that want to send to receiver When you click Rx List you can view all received SMS as follows SMS Rx List Read Status ETS 1 REC READ 006935114545 08 01 01 19 34 22 2 REC READ 885935385852 08 03 12 16 25 27 18 Click the serial no you can view message as follows SMS Reader acted RemotelD 886935366862 08 03 12 162527 Mw Serial can send SMS and receive SMS Back Delete 10 5 use AT Command
4. Ir VobieToLan Settings A Sa NENNT ed Di Lan To Mobile deem Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot O OO Y OC 0 Ek CO SO C The DUMV PCI 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 1 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 Example Lan to Mobile 1 DUMV PCI and Lan 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 DUMV PCI will connect this call auto Example of Application When you call the ch 1 DUMV PCI gsm number it will provide dial tone and you enter a destination number Then ch 2 DUMV PCI will di
5. Register Password Your Voipbuster password Domain Server O Proxy Server 194 221 62 207 Proxy Server s IP _ Outbound Proxy FO Status Reqistered 195 12 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 PORTech Ports Setting Your CTI Partner Route Port of Mobile 1 Mobile SIP Port 5060 1024 65535 TR RTP Port 60000 1024 65535 PERSE Service Domain SIP Port 15062 o 102465535 Port Settings eoo E RTP Port 60100 1024 65535 Ode E 5 Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update system Authority Save Change Reboot 26 12 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 PORTech Codec Settings Your CTI Partner Route Codec Priority Mobile Codec Priority 1 6 711 u law v Network Codec Priority 2 16 711 a law v SIP Settings Codec Priority 3 AES ul Codec Priority 4 G 728 v Service Domain Port Settings Codec Priority 5 G 726 16 wj Codec Settings Codec Priority 8 G726 24 m Codec ID S
6. Web Management fio o Mobile Voip Service Domain Settings You could set information of service domains in this page Route gt Mobile No Mobile 1 y Network Realm 1 Default Active On C Of SIP Settings gt Display Name user 1002 User Ni 1002 NAT Trans M c Register Name 1002 System Auth Register Password Save Change Domain Server 192 168 66 202 Proxy Server 192 168 66 202 Update Outbound Proxy 192 168 66 202 NIRE Status Registered Active On Of Display Name User Nam Register Nami Register Password zl Ea SE 7 73 amp 100 7 pstn gt call 0928492911 mobile number gt DuMV g PCI gt hear the second dial tone call SoftPhone s number gt SoftPhone gt 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 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt Contact lt sip 1002 192 168 66 202 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 9192 168 66 202 CSeq 102 INVITE User Agent Asterisk PBX Max Forwards 70 Date Tue 22 May 2007 02 50 37 GMT Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY 48 Content Type application sdp Content Length 242 v 0 o root 2737 2737 IN IP4 192 168 66 202 s session c IN IP4 192 168 66 202 t 0 0 m audio 1585
7. may enter the whole IP address e g 192 168 0 101 or proxy extension or phone number 2 If this field is blank or simply N it means refuse to transfer 3Jf 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 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 Example 1 Mobile to Lan 0932 0911123456 DuMV PCI have register proxy server Asterisk The proxy server Asterisk have the route 09 When the caller s prefix number is 0932 DuMV QPCI will connect 0911123456 automaticlly 2 Mobile to Lan Any caller call the DuMV QPCI s sim DUMV PCI will prompt dial tone Caller can enter IP or sip extension or phone number sip extension or phone number both need to register SIP Proxy Server or Asterisk Phone number SIP Proxy Server or Asterisk need to set the route of this phone number 9 2 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time DUMV PCI will give priority to Mobile to LAN Speed Dial Settings PORTech Mobile To LAN Speed Dial Your CTI Partner Route Ce NN RE ETE Test 192 168 0 107 O Mobile To Lan Settings Mobile To Lan S
8. via Telnet or your program Allows your program or Telnet Send receive SMS with AT Command Port 23 username voip password x user level 1 command logout module modulel module 2 gt modulel getting module 1 Choose module got press ctrl x to release module 1 Enter ate1 then you can see your at command below Please enter account and password ate1 0 at cmgf 1 at cngs 0911129456 Enter attemgs phone number isst l 0 19 11 Network In Network you can check the Network status configure the WLAN Settings LAN Setting and SNTP settings 11 1 Network Status You can check the current Network setting in this page PO RTech Network Status Your CTI Partner deir Mobile Type Fixed IP Client Fixed IP Client IP 192 168 0 122 192 168 0 102 Network Mask 255 255 255 0 255 255 255 0 Gateway 192 168 0 254 192 168 0 254 WAN Settings MAC 00037 E009999 00037 E008888 LAN Settings SNTP Settings SIP Settings NAT Transform Update System Authority Save Change Reboot 20 11 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 accoun
9. 0037 e008888 DHCP Server DHCP Server OOn QGof Start IP 160 End IP 200 Lease Time 1 lo dd hh 22 11 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 PORTech swrp Settings Your CTI Partner You could set the SNTP servers in this page Route Mobile SNTP On Oof Network BEND Primary Server time windows com Status WAN Settings Secondary Server 208 184 49 9 LAN Settings Time Zone GMT Y 08 vw 00 Y hh mm SIP Settings Sync Time 1 0 0 dd hh mm NAT Transform Update System Authority Save Change Reboot 23 12 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 informations correctly then you can register to SIP Proxy Server correctly 12 1 In Servcie Domain Function you need to input the account and the related informations 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 Acti
10. 03 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 12 headers 0 lines Reliably Transmitting NAT to 192 168 66 203 5060 54 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 lt sip Unknown 192 168 66 202 gt tag as5dee3942 To lt sip 1002 192 168 66 203 5060 gt Contact lt sip Unknown 192 168 66 202 gt Call ID 5ebc2211278e2cb769991 1ad39454d4e 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 Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK672fa67f59c2223275f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Expires 300 Contact lt sip 1002 192 168 66 203 5060 gt expires 300 Date Tue 22 May 2007 03 11 54 GMT Content Length 0 55 21 Simple Steps Step 1 Change the Network setting if you need Network network setting Step
11. 13cm x 32 5cm 5 Chart of the device 5 1 Antenna Antenna connector 5 2 SIM Slot 2 Insert second SIM card 5 3 SIM Slot 1 Insert first SIM card 5 4 WAN RJ 45 internet connector gt standard RJ 45 socket connect to HUB 5 5 LAN LAN port It also can be DHCP Server 6 CABLING 6 1 Connect the internet cable from HUB to the WAN connector of the DuMV PCI If you need to stack up more DUMV PCI you can stack up as follows How to sfack up pm LAN WAN em snm m 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 7 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 Login PORTech VoIP Enter your username and password to login VoIP server Username Password TE a Remember last login Enter the username and password for authentication default username voip password 1234 The page follows when the username and password are correct 8 System Information 8 1 When you login the web page you can see the demo system current system information like firmware version company etc in this page 8 2 Also you can see the f
12. 2 Register SIP proxy Server or Asterisk or VoipBuster if you need sip setting service domain Step 3 Set Route request mobile to lan 1 gt it is two stage dialing when mobile call in DuMV QPCI 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 DUMV PCI 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 gt it is two stage dialing when lan phone call in DUMV PCI will provide dial tone and you can enter mobile number 56 2 specific mobile number when lan phone call in DUMV PCI will connect with the specific mobile number auto 3 gt It is 1 stage dialing When lan phone and DuMV QPCI 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 DuMV Q PCI All changes both need to click save and change 57
13. 2 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 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt tag 677373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 9192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x Content Length 254 v 0 071001 4804366 4807851 IN IP4 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101 a rtpmap 0 pcmu 8000 49 a rtpmap 8 pcma 8000 a rtpmap 3 gsm 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv test 2 SoftPhone gt call 1002 gt DuMV g PCI gt hear second dial tone and call pstn gt pstn answer gt 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 29hG4bK4C4315351FC84CA582D14FB8C25F C3BF From user_1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt Contact lt sip 1001 192 168 66 145 7331 gt Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE Proxy Authorization Digest username 1001 realm asterisk nonce 0d
14. 3b2879 response 8aaaaadb5ad53 654bf0a2ab0fa9bb1 18 uri sip 1002 192 168 66 202 algorithm MD5 Max Forwards 70 Content Type application sdp User Agent X Lite release 1105x Content Length 254 v 0 0 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite 50 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 29hG4bK4C4315351FC84CA582D14FB8C25FC3BF received 192 168 66 145 rport 7331 From user 1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt 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 lt sip 1002 192 168 66 202 gt Content Type application sdp Content Length 242 v 0 o root 2737 2737 IN IP4 192 168 66 202 s session c IN IP4 192 168 66 202 t 0 0 m audio 13798 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 5 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 lt SIP read from 192 168 66 203
15. 5060 REGISTER sip 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK590e92b551 233a10a0ae71944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 10 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 3ca93a1e response 4d39ccb0dae64 bb2f1341e9896ac1ea7 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 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae71944c19b5aa rec 52 eived 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt 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 z9hG4bK590e92b551233a10a0ae7 1944c19b5aa rec eived 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192
16. DUMVAPCI 2 ports GSM VoIP PCI Card User Manual PORTech Communications Inc Content INTRODUCTION ete 1 2 pBUNCTION DESCRIPTION csticccccsiecctssscecececebeseccsccecececesecc esssecedecesecdeccesnceceseoececcssaecesedeseacs 1 IPARISTIS Tai 1 4 DIMENSION 13CM X 32 5CM ccccccscscscscscscscscssssssssscsccssscscscssscssscccsesesessseseseseveseseseseseves 2 5 CHART OF THE DEVICE 2 3 eee eoo goo ceto Pe pee eene co cae esee eaa abaco casos se cede eod o ad SE Re d Ra SERE 3 G CABLING iiie AO 4 7 WEB PAGE SETTING unico sso a eaae aa eee Ta Deae eee saa de eaa a dada 5 S SYSTEM INFORMA TION wscsicccccccccccsccssccscscssocssciscccsccdscoessccseccsesccssossscacocdcccdcocssccssac ecdsceeese 6 9 ROUTE DOSE 6 9 1 MOBILE TO LAN SETTINGS ccccccccccccssssscsccccccccsesssscscecccceeeesssscscccccceeeussscscesccseseassssesess 6 9 2 MOBILE TO LAN SPEED DIAL SETTINGS oooccccccnononnnnnnnncnoncnnononenenococonononnnnnnnncncncnnononannanoss 8 9 3 CALE BACK SERVICE SU SETS a accessus latencia 10 9 4 LAN TO MOBILE SETTINGS ccccccccccccssssssscsccccccceessssscscccccceseessssescccccseseeeesseseccccseseeeeens 11 LOMO 3 FL ovem 13 TO MOBILE S TATUS 2 oor cte A eter erede teneo a eeetede tetas 13 TOZ MOBIEE SETTING e ette A e Re t eO IS 14 10 3 MOBILE FORWARD SETTING u cccccccccccccccccccecccececececccecececececccececeeececececeeeeececesecesesececs 16 10 4 MOBIEE SMS AGENTE 2 estes ehe tee ete e ce eite ete re etae
17. STUN of Mobile 1 O 0n Off STUN of Mobile 2 On Off Route Mobile Network STUN Server SIP Settings STUN Port 3470 1024 65535 NAT Transform Update system Authority Save Change Reboot 33 14 System Auth In System Authority you can change your login name and password PORTeCh system Authority Lace CTI hatapa You could change the login username password in this page Route New username Mobile New password Network Confirmed password SIP Settings NAT Transform Update System Authority Save Change Reboot 34 15 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 PORTech Save Changes Your CTI Partner You have to save changes to effect them Route Mobile cave Changes Network SIP Settings NAT Transform Update New Firmware Default Settings system Authority Save Change Reboot 35 16 Update In Update you can update the system s firmware to the new one or do the factory reset to let the system back to default setting 16 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
18. al this number and connect ch 1 DUMV PCI mobile to lan set route table ch 2 DUMV PCI lan to mobile set route table Additionally two channels DuMV PCI both need to register proxy server or Asterisk And proxy server asterisk set the route that the prefix of destination number dial out from ch 2 DUMV PCI The channel 2 DUMV PCI s ip the first ip 5062 e g http 192 168 0 100 5062 12 10 Mobile 10 1 Mobile Status PO RTech Mobile Status Your CTI Partner 2008 05 16 18 10 Mobile 1 v Mobile Network Registration Chunghwa Seas SIM Card ID o Fwd Settings E SMS Agent Signal Quality 17 Network GSM S N a SIP Settings Incoming IP NAT Transform m Incoming IP Name Update Outgoing IP System Authority Save Change Incoming Mob Reboot Outgoing Mob 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 6 Incoming IP Name proxy server name 7 Outgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call from MOBILE 9 Outgoing Mob The called number of the last outgoing call to MOBILE js 10 2 Mobile Setting PORTech mobile Setting Your CTI Partner Ro
19. an setup the RPort Enable Disable in this page To change this setting please following your ISP information When you finished the setting please click the Submit button PORTech RPort Setting Your CTI Partner Route RPart of Mobile 1 O On O Of Mobile RPort of Mobile 2 90n O Off Network Submit Reset SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 30 12 7 SIP Responses PO RTech SIP Responses Setting Ek CTI Ada Route Response on port busy Mobile 486 Busy here Network 503 Service unavailable _SIP Settings SIP Responses Service Domain SON OOFF 180 Ringing Auto force to ON if 183 was OFF Port Settings OON GOFF 183 Session Progress Codec Settings Codec ID Setting AE Setting Por eiui sn Responses er Settings NAT Transform Update System Authority Save Change Reboot 12 7 1 486 busy here 503 Service unavailable When Device is busy you can select 486 or 505 to response to SIP 12 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 Prompt voice directly For this function 183 must be turn on 12 7 3 183 Session Progress gt It means on progressing When you turn 183 on it means you can hear voicemail
20. dla 18 10 5 USE AT COMMAND VIA TELNET OR YOUR PROGRAM oooocccccononnnnnncncccncnnononnnanacicncnnonononos 19 NETWORK coi eo eoceccc criar 20 12 STP SETTIN CG ui ii 24 13 NAT NO TN 33 I4 S YSTEMCAUTHL i 95 16 edet ica iii 34 ISSAVE CHANGE osorno M n 35 IGUPDA TE ai A AS E 36 LR 1601 87 8 pes 38 IS SPECIEICATICN osito sine depo Ee ER DEI Str ROFR KY paiese REA UERELA TN RIAL DERART UHR kca SEXE ROUEN TAE RR siais 39 19 APPENDIX SETUP DUMV PCI WITH ASTERISK eere eee nnn 40 20 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM DUMY OP Ost eene US la AR 46 21 SIMPLE STEPS mre 56 1 Introduction DuMV PCI 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 enable to make 2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone 2 Function description 2 1 VoIP SIP gt GSM DUMV PCI conversion 2 2 50 sets of LAN gt MOBILE routes setting gt 50 sets of MOBILE gt 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 PC 3 Parts list Please check the parts for any missing parts If do please contact our agents 3 1 DuMV PCI main body 3 2 Network cable 3 3 Antenna 3 4 User Manual 1 3 4 Dimension
21. er 21 How to setup Asterisk to receive Caller ID from DUMV PCI page 42 DuMV PCI will send the message as follows in the Packet From caller number lt sip 3001 192 168 0 228 gt tag 51088abb e Tel Tel DuMV PCI will send the message as follows in the Packet From caller number sip caller number 0192 168 0 228 gt tag 6ac93f7c Please note lf 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 demain User Tel DuMV PCI will send the message as follows in the Packet From Username sip caller number 192 168 0 228 gt tag 7f130947 lt 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 demain 15 8 Presentation CLIR If you need to block the Caller Id for call termination please choose Suppression 9 Mobile PIN Code lf you need to unlock pin code via DuMV g PCl you can 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 12 Answer Delay Delay for incom
22. etting DTMF Setting Codec Priority 7 G 726 32 hl RPort Setting Codec Priority 8 G 726 40 SIP Responses Other Settings RTP Packet Length NAT Transform G 711 amp G 729 20 ms Y Update 6 723 30 ms v System Authority Save Change G 723 5 3K G 723 5 3K On of Reboot Voice VAD Voice VAD OoOn Off 27 12 4 Codec ID Setting You can setup the Codec ID in this page PORTech Codec ID Setting Your CTI Partner You could set the value of Codec ID in this page Route Mobile Codec Type D DefaultValue Network 6726 16 ID 23 85 255 23 6726 24 ID 22 95 255 2 6726 32 ID 2 qu 2 Service Domain A li Y Port Settings 6726 40 ID 21 95 255 21 Codec Settings REC 2633 ID LN 95 255 101 DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot SIP Settings 28 12 5 DTMF Setting You can setup the DTMF Setting in this page PORTech Your CTI Partner Route Mobile Network SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update m System Authority Save Change Reboot DTMF Setting Mobile DTMF Transfer to Lan 9 2833 O Inband DTMF O Send DTMF SIP Info 29 12 6 RPort Function You c
23. ing call when the ring 10 3 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 equipments PORTech Forward Setting Your CTI Partner Route O Forward Enable Mobile Name URL Port Status Fwd to Mobile1 192 168 0 100 5060 tings Fwd to Mobile2 192 168 0 100 5062 Fwd Settings SMS Agent Fwd to External Network SIP Settings NAT Transform Update System Authority Save Change Reboot d Forward Enable is not motivate on Defualt value 16 So please mark Forward Enable this blank to motivate this function Take SJ Phone for example Profiles gt Edit gt Advanced gt Accept redirection replies Turn on the Forward Enable therefore the SJ Phone can designate a port which are free to use Profile Options General itializati SIP Proxy Advanced Use short headers Expose software version Use obsolete transfer mechanism BY E Also Restrict caller identity support varies for proxies from a different vendors a Use standard status messages otherwise messages will be taken from SIP packets Voice mail number or address Y Remove fancy characters from phone numbers E Name URL Port Fwd to Mobile1 monos 7 0 Fwd to Mobile2 peoos Fwd to External O The Explanation of Picture
24. ings NAT Transform Update System Authority Save Change 38 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 723 1 5 3k G 723 1 6 3k G 729A G 729A B 18 4 Voice Quality VAD 230 CNG AEC LEC Packet loss 18 5 GSM DUMV PCI Dual BAND 900 1800 MHZ Tri BAND BenQ M23 900 1800 1900 MHZ Tri BAND Siemens MC56 850 1800 1900 MHZ Quad BAND 900 1800 1900 850 MHZ 19 Appendix Setup DuMV PCI with Asterisk 19 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 lt gsm network gt DUMV PCI lt an gt Asterisk lt internet gt VOIP provider lt whatever gt landline To do such a call you just call your DUMV PCI number it has its own simcard 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 DUMV PCI for free You can then call all around the world from your mobile at voip cost 19 2 DUMV PCI 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 int
25. l the DUMV PCI sim card mail box thru GSM exten gt _888 1 SetCallerl D xXxxxxxxxxx exten gt _888 2 Dial SIP EXTEN 103 60 r exten gt _888 3 Hangup 45 20 How to setup Asterisk to receive Caller ID from DuMV PCI Test version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a X Lite 1105x Modify file e 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 secret 1002 qualify yes 46 nat yes host dynamic canreinvite no context internal Add the following setting to etc asterisk extensions conf internal exten gt 1000 1 Dial SIP 1000 exten gt 1001 1 Dial SIP 1001 exten gt 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 DUMV PCI address 192 168 66 203 5060 username 1002 displayname user_1002 47 VoIP Web Management Windo Explorer Windows Internet Exp A g gt Je hitp 7192 168 66 203 ogin E ES x five Search y Pl BRO SEO VAD SSB TAO HAG mene cpuem gt ae abala SE SEES E VoIP
26. o account To have the DuUMV PCI to work with Asterisk you need first to 40 configure the box Here are some screen shots showing all the important parameters You have to note that in all the configuration process the DuMV g PCI is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation WAN Settings You could configure the WWAN settings in this page WAN Setting IP Type O Fixed IP C DHCP Client C PPPoE IP M V37D IP Mask 255 255 255 0 i Gateway Router iP 7 DNS Serverl 168 95 192 1 DNS Server2 1168 95 1 1 MAC PPPoE Setting User Name Password Submit Reset Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM LAN To Mobile Table Page 1 a Your Asterisk IP 4 dl 1 3 4 6 i 8 3 4 Mobile To LAN Table Page 1 he tem E O A CO Select 0 Authorised Mobile 103 1 Another Authorised Mobile 103 2 3 4 5 6 7 8 S 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 Any number which is not in that list won t have acces to the LAN side so to Asterisk If you want to allow an
27. operly 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 the signal quality down to 11 audio becomes very jerky So maximum signal quality maximum audio quality 19 4 Asterisk configuration Once the DuMV g PCI is set you have to configure Asterisk On that side you have to setup files as follow 19 5 sip conf GSM VOIP Gateway DUMV PCI 103 type friend 44 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 lt 103 gt 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 19 6 extensions conf GSM Gateway incoming calls gateway exten gt 103 1 Answer exten gt 103 2 DigitTimeout 3 give enough time to do second stage dialing exten gt _103 3 ResponseTimeout 5 exten gt 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 cal
28. peed Dia Umane d UD L Mobile Network SIP Settings NAT Transform Update System Authority Save Change Delete Selected Delete All Reboot 0D 0 100 hh C hl O 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 Num 0 Name test URL 192 168 0 107 When the caller hear dial tone and enter 0 system will connect 192 168 0 107 9 3 Call Back Service 50 sets PORTech Mobile To LAN Table bi CTI Zend Route Page Y cin URL 0933579613 O 886933579613 O Mobile To Lan S Speed Dial Lan To Mobile Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot wo Do A OY GOV dx O NN CO Delete Selected Delete All Add New Position 0 49 CID Ex 0911111111 0911 URL Ex 192 168 0 1 28t You can set call back service as the following steps 1 CID set the phone number here up to 50 sets 2 URL is the command of call back Application a Call MV 370 b MV 370 will detect the phone number is in call back list or not c If yes MV 370 will reject the call and call it back d You will receive the call from MV 370 and prompt a dial tone 10 9 4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE PORTech LAN To Mobile Table bh CTI E Route Page
29. t from your Service Provider please input the Username and the Password correctly 3 The Bridge Item is to setuo 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 PO RTech WAN Settings Your CTI Partner You could configure the WAN settings in this page Route Network Mode OBridge 9 NAT Mobile Network WAN Setting IP Type O Fixed IP O DHCP Client O PPPoE WAN Settings IP 192 168 0 122 LAN Settings Mask 255 255 255 0 SNTP Settings SIP Petes DNS Severi 168 95 192 1 NAT Transform DNS Server2 168 95 1 1 Update MAC 00037 e009999 System Authority Save Change PPPoE Setting Reboot User Name Password E 21 11 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 PORTech Your CTI Partner Route Mobile Network Status WAN Settings LAN Settings SNTP Settings SIP Settings NAT Transform Update System Authority Save Change Reboot LAN Settings LAN Setting IP 1192 168 0 102 Mask 255 255 255 MAC 10
30. unction lists in the left side You can use mouse to click the function you want to set up PORTech Mobile VoIP 6 691a Your CTI Partner Route Model Name MV 372 Mobile Model Description GSM 900 1800MHz HO Firmware Version Fri May 16 11 30 35 2008 Codec Version Mon Jul 24 10 55 05 2006 SIP Settings NAT Transform Update System Autharity 2007 PORTech Communications Inc Save Change Reboot 9 Route 9 1 Mobile TO LAN Settings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN PORTech mobile To LAN Table Your CTI Partner Route Page Y Mobile To Lan Speed Dip O Lan To Mobile Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot no ON C C1 fF C l0 O Delete Selected Delete All Add New Pasition 0 49 CID Ex 0911111111 0911 URL Ex 192 168 0 1 28t The DuMV gPCI 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 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
31. ute 1 WolP Tx Gain E 0 12 2 VoIP Rx Gain 11 0 15 Mobile 3 LAN Dialtone Gain E 0 12 Settings Fwd Settings 4 Routing Range D to 49 D 49 SMS Agent 5 CODEC Tx Gain 6 7 6 CODEC Rx Gain 6 7 Sennen 7 SIP From Tel User Standard Answer Delay D 0 15 12 SIP Settings 8 CLID Presentation Suppression Invocation NAT Transform 9 Mobile PIN Code On E Code Confirmed Update LAN Answer Mode Answered Alerted Income System Authority i vu A Reboot Routing Range o to 49 D 49 CODEC Tx Gain e 0 7 CODEC Rx Gain e 0 7 SIP From Tel User Standard Answer Delay 0 0 15 CLID Presentation Suppression Invocation Mobile PIN Code On Code Confirmed LAN Answer Mode Answered Alerted Income Mobile 1 E DVoIP Tx Gain Mobile 2 Rx 4 2 VoIP Rx Gain 1 VoIP Tx Gain To adjust the volume of LAN side 14 2 VoIP Rx Gain To adjust the volume of Mobile side 3 LAN Dialtone Gain DTMF Reciver is not good you can adjust gain down 4 ON Off If you use this channel please click on Otherwise please click off B CODEC Tx Gain as above 6 CODEC Rx Gain as above 7 SIP From Caller ID transfer e 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 ref
32. ve 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 24 PORTech MEE Service Domain Settings Route Mobile 1 v ee Network Active O ON O OFF SIP Settings Display Name 3001 aln Por Settings Register Narne 3001 EE E Register Password esco Codec ID Setting DTMF Setting Domain Server RPort Setting Proxy Server 61 218 151 230 SIP Responses Other Settings Outbound Proxy NAT Transform Status Not Registered Example Register VoipBuster Realm 1 Default Active On C of Display Name fenyo User Name fienny0922 Your Voipbuster username Register Name emny0922
33. while GMS side is busy We recommend you to turn this on if you use SIP Proxy 31 12 8 Other Settings Other Settings you can setup the Hold by RFC and QoS in this page To change these settings please following 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 PORTech Other Settings Your CTI Partner Route Hold by RFC of Mobile 1 OO Off Hold by RFC of Mobile 2 O O0n Off Mobile Network Voice QoS 40 0 63 SIP Settings SIP QoS 40 083 Service Domain SIP Expire Time 30 60 86400 sec Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 32 13 NAT Trans In NAT Trans you can setup STUN and uPnP function These functions can help your VoIP device working properly behind NAT 13 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 PORTech STUN Setting E CT El
34. y number just set in that field but beware of the bill 42 Service Domain Settings Realm 1 Default Active O ON O OFF Display Name 103 m User Name 103 Register Name 1103 Register Password Domain Server Asterisk IP Proxy Server Outbound Proxy Status Not Registered Once Asterisk configuration is made you should get Registered on the Realm1 Codec Settings Codec Priority Codec Priority 1 G 711 u law Codec Priority 2 G 711 a law v Codec Priority 3 Mot Used v Codec Priority 4 Not Used v Codec Priority 5 Not Used 3 Codec Priority 6 Not Used v Codec Priority 7 Not Used Ww Codec Priority 8 Mot Used Ww RTP Packet Length G 711 amp G 729 20 ms r2 2g 30 ms G 723 5 3K if 2o ES O On of Voice VAD Voice VAD O On Off 43 It is very important to use only u law or a law 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 Mobile Setting VoIP Tx Gain HO 1 12 VoIP Rx Gain 3 0 15 LAN Dialtone Gain 10 0 12 Mobile ON OFF Routing Range lto 49 0 49 CODEC Tx Gain 6 0 7 CODEC Rx Gain B 0 7 SIP From Tel User Standard Answer Delay 10 0 15 CLID Presentation Suppression Invocation These settings seem to be ok just adjust 19 3 Antenna position Another important thing is to pr
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