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Portech DuMV@PC 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.
VoIPon www.voipon.co.uk sales@voipon.co.uk
Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299

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  • Page 1 DuMV@PCI 2 ports GSM/VoIP PCI Card User Manual PORTech Communications Inc. VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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 VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • Page 4 VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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. 2.Function description 2.1 VoIP(SIP)、GSM(DuMV@PCI) conversion.

  • Page 6: Dimension: 13Cm X 32.5Cm

    4.Dimension: 13cm x 32.5cm VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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. VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • Page 8: Cabling

    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. 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 (3) * means all numbers can be accepted (4) N means the calls without the CID Please note the priority of the rules.

  • Page 12: Mobile To Lan Speed Dial Settings

    *URL：The IP address to transfer this call (1) 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. (3) If an ‘*’ entered, it means 2-stages-dialing.
  • 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 VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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. may enter the whole IP address, e.g.
  • Page 16 *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.

  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • Page 19 (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. (5)CODEC Tx Gain: as above (6)CODEC Rx Gain: as above (7) SIP From: Caller ID transfer Tel/User(Standard): If you need to register to Asterisk and proxy...

  • 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 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...
  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk...
  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • Page 32 12.4 Codec ID Setting You can setup the Codec ID in this page. -28- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • Page 33 12.5 DTMF Setting You can setup the DTMF Setting in this page. -29- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

  • 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...

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

    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 <----gsm network---->...
  • Page 45 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@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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • Page 48 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. These settings seem to be ok, just adjust ...
  • Page 49 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 <103> host=dynamic nat=no ; Gateway is not behind a NAT router canreinvite=no ;...

  • 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 canreinvite=no context=internal [1002]...
  • Page 51 nat=yes host=dynamic canreinvite=no context=internal 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- VoIPon www.voipon.co.uk sales@voipon.co.uk Tel: +44 (0)1245 808195 Fax: +44 (0)1245 808299...
  • Page 52 test1 pstn call 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> Contact: <sip:1002@192.168.66.202>...
  • 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 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 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 C3BF From: user_1001 <sip:1001@192.168.66.202:7331>;tag=1121869743...
  • 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

    21. Simple Steps Step 1. Change the Network setting if you need (Network/network setting) Step 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) *,* --->it is two stage dialing.
  • 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. * Please note:Asterisk need to set route of destination number that dial out from DuMV@PCI * All changes both need to click "save and change"...

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