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Pulse DuMV@PCI User Manual

2 ports gsm/voip pci card
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DuMV@PCI
2 ports GSM/VoIP PCI Card
User Manual
PORTech Communications Inc.

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Summary of Contents for Pulse DuMV@PCI

  • Page 1 DuMV@PCI 2 ports GSM/VoIP PCI Card User Manual PORTech Communications Inc.

  • Page 2: Table Of Contents

    【Content】 1.INTRODUCTION......................... 1 2.FUNCTION DESCRIPTION....................1 3.PARTS LIST.......................... 1 4.DIMENSION: 13CM X 32.5CM..................2 5.CHART OF THE DEVICE....................3 6.CABLING ..........................4 7.WEB PAGE SETTING......................5 8.SYSTEM INFORMATION....................6 9. ROUTE..........................6 9.1 M TO LAN S ..................... 6 OBILE ETTINGS 9.2 M LAN S...
  • Page 3 16.UPDATE ..........................36 17.REBOOT..........................38 18.SPECIFICATION ......................39 19. APPENDIX: SETUP DUMV@PCI WITH ASTERISK ..........40 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM DUMV@PCI ........................46 21. SIMPLE STEPS ....................... 56...

  • Page 5: Introduction

    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.

  • Page 6: Dimension: 13Cm X 32.5Cm

    4.Dimension: 13cm x 32.5cm...

  • Page 7: Chart Of The Device

    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，standard RJ-45 socket，connect to HUB. 5.5 LAN：LAN port. It also can be DHCP Server.

  • Page 8: 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. 6.2 Connect the antenna and put it in proper position to get the best signal reception.

  • Page 9: Web

    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： Enter the username and password for authentication. (default username=voip, password=1234).

  • Page 10: System Information

    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 function lists in the left side. You can use mouse to click the function you want to set up.
  • Page 11 The DuMV@PCI 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...

  • Page 12: Mobile To Lan Speed Dial Settings

    *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.
  • Page 13 *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...

  • Page 14: Call Back Service (50 Sets)

    9.3 Call Back Service (50 sets) 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.

  • Page 15: Lan To Mobile Settings

    9.4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call 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.
  • Page 16 (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 dial this number and connect.

  • Page 17: Mobile

    10.Mobile 10.1 Mobile Status (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.

  • Page 18: Mobile Setting

    10.2 Mobile Setting (12) (10) Mobile 1: (6)Rx VoIP Codec (5) Tx DTMF Mobile 2: (1)VoIP Tx Gain Codec (2) VoIP Rx Gain DTMF (1) VoIP Tx Gain: To adjust the volume of LAN side. -14-...
  • Page 19 And how to transfer the caller ID to LAN,please refer 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 " <sip:3001@192.168.0.228>;tag=51088abb Tel/Tel : DuMV@PCI will send the message as follows in the Packet.

  • Page 20: Mobile / Forward Setting

    (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 DuMV@PCI,you can click “On” and enter pin code. (10)LAN Answer Mode: Answered : when mobile answer,then connect the call...
  • Page 21 So please, mark "Forward Enable" this blank to motivate this function. Take SJ Phone for example: Profiles -> Edit -> Advanced -> Accept redirection replies (Turn on the "Forward Enable", therefore the SJ Phone can designate a port which are free to use.) Name URL:Port 192.168.0.100:5060...

  • Page 22: Mobile / Sms Agent

    10.4 Mobile / SMS Agent : Read received SMS (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. -18-...

  • Page 23: Use At Command Via Telnet Or Your Program

    Click the serial no,you can view message as follows. 10.5 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 command below Enter at+cmgs=”phone number”...

  • Page 24: Network

    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. -20-...
  • Page 25 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.
  • Page 26 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 -22-...
  • Page 27 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.

  • Page 28: Sip Setting

    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.
  • Page 29 Example: Register VoipBuster Your Voipbuster username Your Voipbuster password Proxy Server’s IP -25-...
  • Page 30 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.
  • 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 button. -27-...
  • Page 32 12.4 Codec ID Setting You can setup the Codec ID in this page. -28-...
  • Page 33 12.5 DTMF Setting You can setup the DTMF Setting in this page. -29-...
  • Page 34 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. -30-...
  • Page 35 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.
  • Page 36 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’...

  • Page 37: Nat Trans

    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.

  • Page 38: System Auth

    14.System Auth. In System Authority you can change your login name and password. -34-...

  • Page 39: Save Change

    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.

  • Page 40: Update

    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.
  • Page 41 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. -37-...

  • Page 42: Reboot

    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. -38-...

  • Page 43: Specification

    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 -39-...

  • Page 44: Appendix: Setup Dumv@Pci With Asterisk

    Your mobile <----gsm network----> DuMV@PCI <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> 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.
  • Page 45 Here are some screen shots showing all the important 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 important to avoid the two stage dialing when you give a call from Asterisk to GSM.
  • Page 46 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 47 Once Asterisk configuration is made, you should get 'Registered' on the Realm1. -43-...
  • Page 48 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@PCI is set, you have to configure Asterisk. On that side, you have to setup files as follow : 19.5 sip.conf ;...
  • Page 49 => _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 DuMV@PCI 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 50: How To Setup Asterisk To Receive Caller Id From Dumv@Pci

    How to setup Asterisk to receive Caller ID from DuMV@PCI 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...
  • Page 51 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 DUMV@PCI: address=192.168.66.203:5060; username=1002, displayname=user_1002 -47-...
  • Page 52 0928492911(mobile number) DuMV@PCI 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 53 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 54 0-15 a=sendrecv test 2 SoftPhone call 1002 DuMV@PCI 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 55 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 56 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 57 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 58 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 59 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 Via: SIP/2.0/UDP...

  • Page 60: Simple Steps

    (1) *,* --->it is two stage dialing. when mobile call in, DuMV@PCI 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 61 (2) *, specific mobile number when lan phone call in, DuMV@PCI will connect with the specific 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.

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