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Portech MV-370 User Manual

Voip gsm gateway
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MV-370 / MV-372
VoIP GSM Gateway

User Manual

MV-370
MV-372
PORTech Communications Inc.

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Summary of Contents for Portech MV-370

  • Page 1: User Manual

    MV-370 / MV-372 VoIP GSM Gateway User Manual MV-370 MV-372 PORTech Communications Inc.

  • Page 2: Table Of Contents

    15.SYSTEM AUTH......................... 36 16.SAVE CHANGE......................... 37 17.UPDATE ..........................38 18.REBOOT..........................40 19. IP SETTING........................41 20.SPECIFICATION ....................... 43 21. APPENDIX: SETUP MV-370/MV-372 WITH ASTERISK..........44 22.HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM MV-370/MV-372..50 23. SIMPLE STEPS ......................... 60...

  • Page 3: Introduction

    1.Introduction MV-370/MV-372 is a 1/2 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 1/2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone.

  • Page 4: Dimension:14.5Cm X 17Cm X 3.9Cm

    (3.1) MV-370 (3.1) MV-372 4.Dimension:14.5cm x 17cm x 3.9cm 3.9cm 17cm 14.5cm...

  • Page 5: Mv-370 Panel Description

    5. MV-370 Panel description 14.5cm 3.9cm 17cm (5.1) (5.2) (5.3) (5.4) (5.6) (5.5) (5.7) 5.1 Antenna：Antenna connector. 5.2 DC 12V：Power socket. 5.3 LAN: Standard RJ-45 socket, connecting to Hub circuit. 5.4 PWR: Power indicator light, red light. Light is on when system’s power supply is normal.

  • Page 6: Mv-372 Panel Description

    6. MV-372 Panel description 6.5 6.6 6.7 6.8 6.1 Antenna：Antenna connector. 6.2 DC 12V：Power input. 6.3 LAN：LAN port. It also can be DHCP Server. 6.4 WAN: RJ-45 internet connector，standard RJ-45 socket，connect to HUB. 6.5 PWR (Power LED)：Light up when power is normal. 6.6 VoIP1：an indicator light of VoIP1 6.7 VoIP2：an indicator light of VoIP2 6.8 LINK Indicator：Light up when network is connected.

  • Page 7: Cabling

    7.3 Insert the SIM card from back of the main body. (take the slide off first). Click reset button 3 sec. MV-370/MV-372 will restore default IP. Other setting as usual. 7.5 Connect the power adaptor. The ‘POWER’ LED should be light up.

  • Page 8: Web

    8.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 9: System Information

    9.System Information. 9.1 When you login the web page, you can see the demo system current system information like firmware version, company… etc in this page. 9.2 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 10: Mobile To Lan Settings

    10.1 Mobile TO LAN Settings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN. The MV-370/MV-372 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.
  • Page 11 When the caller’s prefix number is 0932,MV-370/MV-372 will connect 0911123456 automaticlly (2) Mobile to Lan: *,* Any caller call the MV-370/MV-372’s sim,MV-370/MV-372 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.

  • Page 12: Call Back Service (50 Sets) **New Feature

    Application: a. Call MV-370/ MV-372 b. MV-370/ MV-372 will detect the phone number is in call back list or not c. If yes, MV-370/ MV-372 will reject the call, and call it back d. You will receive the call from MV-370/ MV-372, and prompt a dial tone...

  • Page 13: Mobile To Lan Speed Dial Settings

    10.3 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time,MV-370/MV-372 will give priority to Mobile to LAN Speed Dial Settings. *The call will be answered and prompt dial tone again. When the caller may enter the “Num”, system will connect the “URL”...

  • Page 14: Lan To Mobile Settings

    10.4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE. The MV-370/MV-372 will transfer to the mobile number according to the incoming URL *URL：The IP address of the incoming call.
  • Page 15 (3)When you dial any destination phone number from lan phone,MV-370/MV-372 will connect this call auto. Example of Application: When you call the ch.1 MV-370/MV-372 gsm number,it will provide dial tone and you enter a destination number. Then ch.2 MV-370/MV-372 will dial this number and connect.

  • Page 16: Mobile

    11.Mobile 11.1 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 17: Mobile Setting

    11.2 Mobile Setting (12) (10) (11) 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. -15-...
  • Page 18 And how to transfer the caller ID to LAN,please refer 21.How to setup Asterisk to receive Caller ID from MV-370/MV-372 (page 42) MV-370/MV-372 will send the message as follows in the Packet. From: " caller number " <sip:3001@192.168.0.228>;tag=51088abb Tel/Tel : MV-370/MV-372 will send the message as follows in the Packet.

  • Page 19: Mobile / Forward Setting

    Code:If need unlock code MV-370/MV-372,you can click “On” and enter pin code. (11)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 (12)Answer Delay: Delay for incoming call when the ring.
  • Page 20 "Forward Enable" is not motivate on Defualt value. 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.) -18-...
  • Page 21 Name URL:Port 192.168.0.100:5060 Fwd to Mobile1: 192.168.0.100:5062 Fwd to Mobile2: 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 (192.168.0.100). 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).

  • Page 22: Mobile / Sms Agent

    11.4 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. Click the serial no,you can view message as follows.

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

    11.5 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” Enter short message -21-...

  • Page 24: Network

    12.Network In Network you can check the Network status, configure the WLAN Settings , LAN Setting and SNTP settings. 12.1 Network Status: You can check the current Network setting in this page. 12.2 WAN Settings: You can check the current Network setting in this page.
  • Page 25 12.3 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 -23-...
  • Page 26 -24-...
  • Page 27 12.4 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

    13.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. 13.1 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 -27-...
  • Page 30 13.2 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 13.3 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. -29-...
  • Page 32 13.4 Codec ID Setting You can setup the Codec ID in this page. -30-...
  • Page 33 13.5 DTMF Setting You can setup the DTMF Setting in this page. -31-...
  • Page 34 13.6 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. -32-...
  • Page 35 13.7 SIP Responses 13.7.1 486(busy here), 503(Service unavailable): When Device are busying, you can select 486 or 505 to response to SIP. 13.7.2 180 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 Voice-Mail directly.
  • Page 36 13.8 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

    14. NAT Trans In NAT Trans. you can setup STUN and uPnP function. These functions can help your VoIP device working properly behind NAT. 14.1 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

    15.System Auth. In System Authority you can change your login name and password. -36-...

  • Page 39: Save Change

    16.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

    17.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. 17.1 Update firmware (1) In New Firmware function you can update new firmware via HTTP in this page.
  • Page 41 17.2 Restore Default Settings In this page: Update/ Default Settings, you could restore the factory default settings to the system. All setting will restore default setting. IP will retain original IP as usual not default IP. -39-...

  • Page 42: Reboot

    18.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. -40-...

  • Page 43: Ip Setting

    19. IP Setting The operator can setup or query the network parameters by dialing in the mobile number which it SIM card has been put in the main body. The status or result is response by voice. In the first 20 seconds after power-on, the VoIP GSM Gateway enters the IP setting mode.
  • Page 44 setting in the Primary DNS DNS Server field. Default : 192.168.0.1 IVR will announce the version Check Firmware #128# of the firmware running Version The system will change to DHCP #111# DHCP client Client type DHCP will be disabled and Set Static IP #112xxx*xxx*xxx system will change to the...

  • Page 45: Specification

    20.Specification 20.1 Protocols SIP (RFC2543,RFC3261) 20.2 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 20.3 Codec G.711 u-Law G.711 a-Law G.723.1 (5.3k) G.723.1 (6.3k) G.729A G.729A/B 20.4 Voice Quality -43-...

  • Page 46: Appendix: Setup Mv-370/Mv-372 With Asterisk

    Your mobile <----gsm network----> MV-370/MV-372 <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> landline To do such a call, you just call your MV-370/MV-372 number (it has its own simcard), then you get an invitation tone, then you dial the number which is handled by Asterisk.
  • Page 47 Here are some screen shots showing all the important parameters. You have to note that in all the configuration process, the MV-370/MV-372 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 48 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 49 Once Asterisk configuration is made, you should get 'Registered' on the Realm1. -47-...
  • Page 50 On the other end,the signal quality down to 11, audio becomes very jerky. So, maximum signal quality = maximum audio quality. 21.4 Asterisk configuration Once the MV-370/MV-372 is set, you have to configure Asterisk. On that side, you have to setup files as follow : 21.5 sip.conf ;...
  • Page 51 => _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 MV-370/MV-372 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 52: How To Setup Asterisk To Receive Caller Id From Mv-370/Mv-372

    22.How to setup Asterisk to receive Caller ID from MV-370/MV-372 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 53 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 MV-370/MV-372: address=192.168.66.203:5060; username=1002, displayname=user_1002 -51-...
  • Page 54 0928492911(mobile number) MV-370/MV-372 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 55 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 56 0-15 a=sendrecv test 2 SoftPhone call 1002 MV-370/MV-372 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 57 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 58 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 59 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 60 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 61 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 62: Simple Steps

    (1) *,* --->it is two stage dialing. when mobile call in,MV-370/MV-372 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 63 15.21 Federal Communications Commission (FCC) Statement You are cautioned that changes or modifications not expressly approved by the part responsible for compliance could void the user’s authority to operate the equipment. 15.105(b) Federal Communications Commission (FCC) Statement This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC rules.
  • Page 64 FCC RF Radiation Exposure Statement: This Transmitter must not be co-located or operating in conjunction with any other antenna or transmitter. This equipment complies with FCC RF radiation exposure limits set forth for an uncontrolled environment. This equipment should be installed and operated with a minimum distance of 20 centimeters between the radiator and your body.

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Mv-372

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