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12.7 SIP Responses

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)-->[It means"on progressing"] : When you

turn 183 on, it means you can hear voicemail while GMS side is

busy. We recommend you to turn this on if you use SIP Proxy.

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Summary of Contents for DuMV@PCI

Page 1: ...DuMV PCI 2 ports GSM VoIP PCI Card User Manual PORTech Communications Inc...

Page 2: ...6 9 1 MOBILE TO LAN SETTINGS 6 9 2 MOBILE TO LAN SPEED DIAL SETTINGS 8 9 3 CALL BACK SERVICE 50 SETS 10 9 4 LAN TO MOBILE SETTINGS 11 10 MOBILE 13 10 1 MOBILE STATUS 13 10 2 MOBILE SETTING 14 10 3 MOB...

Page 3: ...16 UPDATE 36 17 REBOOT 38 18 SPECIFICATION 39 19 APPENDIX SETUP DUMV PCI WITH ASTERISK 40 20 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM DUMV PCI 46 21 SIMPLE STEPS 56...

Page 4: ......

Page 5: ...unction description 2 1 VoIP SIP GSM DuMV PCI 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...

Page 6: ...2 4 Dimension 13cm x 32 5cm 1 3 2...

Page 7: ...enna 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 standard RJ 45 socket connect to HUB 5 5 LAN LAN port It also can be...

Page 8: ...f you need to stack up more DuMV PCI 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 bo...

Page 9: ...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 vo...

Page 10: ...rmation like firmware version company etc in this page 8 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 9 Route 9 1 Mobile TO LAN Set...

Page 11: ...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 r...

Page 12: ...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 pre...

Page 13: ...ered 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 wil...

Page 14: ...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 t...

Page 15: ...ll incoming from LAN to MOBILE 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...

Page 16: ...d 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 thi...

Page 17: ...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 cal...

Page 18: ...14 10 2 Mobile Setting 1 VoIP Tx Gain To adjust the volume of LAN side GSM VoIP Codec LAN 6 Rx 5 Tx DTMF 1 VoIP Tx Gain 2 VoIP Rx Gain 1 2 3 5 6 7 8 9 10 GSM Codec Rx Tx DTMF Mobile 1 Mobile 2 4 12...

Page 19: ...he message as follows in the Packet From caller number sip 3001 192 168 0 228 tag 51088abb Tel Tel DuMV PCI will send the message as follows in the Packet From caller number sip caller number 192 168...

Page 20: ...en 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...

Page 21: ...Fwd to Mobile2 192 168 0 100 5062 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...

Page 22: ...Agent 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 follow...

Page 23: ...5 use AT Command via Telnet or your program Allows your program or Telnet Send receive SMS with AT Command Port 23 Please enter account and password Choose module Enter ate1 then you can see your at c...

Page 24: ...20 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...

Page 25: ...re 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 co...

Page 26: ...his 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 t...

Page 27: ...u 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 finis...

Page 28: ...k 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 th...

Page 29: ...25 Example Register VoipBuster Your Voipbuster username Your Voipbuster password Proxy Server s IP...

Page 30: ...an 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 ple...

Page 31: ...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 but...

Page 32: ...28 12 4 Codec ID Setting You can setup the Codec ID in this page...

Page 33: ...29 12 5 DTMF Setting You can setup the DTMF Setting in this page...

Page 34: ...30 12 6 RPort Function You can 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...

Page 35: ...LE 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 Sess...

Page 36: ...llowing 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...

Page 37: ...king 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 t...

Page 38: ...34 14 System Auth In System Authority you can change your login name and password...

Page 39: ...ange 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 an...

Page 40: ...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...

Page 41: ...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 rest...

Page 42: ...38 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...

Page 43: ...61 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 3...

Page 44: ...tever 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 specia...

Page 45: ...parameters You have to note that in all the configuration process the DuMV PCI is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation Here the is importan...

Page 46: ...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 47: ...43 Once Asterisk configuration is made you should get Registered on the Realm1...

Page 48: ...rovided 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 o...

Page 49: ...f 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 _103 1 Answer exten _103 2 Dig...

Page 50: ...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 secre...

Page 51: ...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 a...

Page 52: ...92 168 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 1...

Page 53: ...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 54: ...ch 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 C...

Page 55: ...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 56: ...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 57: ...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 58: ...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 1...

Page 59: ...ITE 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 z9hG4bK672fa67f59c2223275f5ee286d27...

Page 60: ...er 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...

Page 61: ...mobile number auto 3 It is 1 stage dialing When lan phone and DuMV PCI both register Asterisk you can dial any destination number from lan phone directly Please note Asterisk need to set route of des...

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