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Portech MV-374 User Manual

Voip gsm gateway
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MV-374 / MV-378
VoIP GSM Gateway

User Manual

MV-374
MV-378
PORTech Communications Inc.

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Summary of Contents for Portech MV-374

  • Page 1: User Manual

    MV-374 / MV-378 VoIP GSM Gateway User Manual MV-374 MV-378 PORTech Communications Inc.

  • Page 2: Table Of Contents

    11.SIP SETTING ........................25 12. NAT TRANS ........................34 13.SYSTEM AUTHORITY ....................35 14.UPDATE..........................36 15.SAVE CHANGE ........................38 16.REBOOT ..........................39 17. IP SETTING ........................40 18.SPECIFICATION ......................42 19. APPENDIX: SETUP MV-374/MV-378 WITH ASTERISK.........43 20.HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM MV-374/MV-378 ..............................50 21. SIMPLE STEPS........................59...

  • Page 3: Introduction

    1.Introduction MV-374/MV-378 is a 4 / 8 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 4 / 8 calls simultaneously from IP phones to GSM networks and GSM network to IP phone.

  • Page 4: Dimension : 30X28X4 Cm

    (3.1) (3.1) MV-378 MV-374 (3.2) MV-378 (3.2) MV-374 (3.4) (3.3) (3.5)-option 4.Dimension : 30x28x4 cm...

  • Page 5: Chart Of The Device

    5.Chart of the device 5.5 5.6 5.7 5.1 Antenna：Antenna connector. 5.2 WAN: RJ-45 internet connector，standard RJ-45 socket，connect to HUB. 5.3 DC 12V：Power input. 5.4 SIM Card 5.5 LINK Indicator：Light up when network is connected. 5.6 CH3：an indicator light of VoIP3 5.7 CH4：an indicator light of VoIP4 5.8 PWR (Power LED)：Light up when power is normal.

  • Page 6: Web

    6.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 7: System Information

    7.System Information. 7.1 When you login the web page, you can see the demo system current system information like firmware version, company… etc in this page. 7.2 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 8 8.1 Mobile TO LAN Settings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN. The MV-374/MV-378 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 9 (1) Mobile to Lan: 0932*,0911123456 MV-374/MV-378 have register proxy server/Asterisk The proxy server/Asterisk have the route “09” When the caller’s prefix number is 0932,MV-374/MV-378 will connect 0911123456 automaticlly (2) Mobile to Lan: *,* Any caller call the MV-374/MV-378’s sim,MV-374/MV-378 will prompt dial tone.Caller can enter IP or sip extension or phone...
  • Page 10 8.2 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time,MV-374/MV-378 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 11 8.3 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE. The MV-374/MV-378 will transfer to the mobile number according to the incoming URL *URL：The IP address of the incoming call.
  • Page 12 (3)When you dial any destination phone number from lan phone,MV-374/MV-378 will connect this call auto. Example of Application: When you call the ch.1 MV-374/MV-378 gsm number,it will provide dial tone and you enter a destination number. Then ch.2 MV-374/MV-378 will dial this number and connect.
  • Page 13 MV-374/MV-378’s IP: The channel 1:192.168.0.100:5060 The channel 2:192.168.0.100:5062 The channel 3:192.168.0.102:5060 The channel 4:192.168.0.102:5062 The channel 5:192.168.0.104:5060 The channel 6:192.168.0.104:5062 The channel 7:192.168.0.106:5060 The channel 8:192.168.0.106:5062 -11-...

  • Page 14: Mobile

    9.Mobile 9.1 Mobile Status (1)Choose Mobile 1,2,3 or 4 (MV-378: Mobile 1,2,3,4,5,6,7,8) (2)Network Registration：The telecom carrier which the SIM card been registered. (3)SIM Card ID：SIM card ID. (4)Signal Quality：Signal quality. (5)GSM S/N : IMEI Number (6)Incoming IP：The IP address of the last incoming call from LAN. (7)Incoming IP Name: proxy server name (8)Outgoing IP：The IP address of the last outgoing call to LAN.

  • Page 15: Mobile Setting

    9.2 Mobile Setting (10) (11) (12) Mobile 1: (6)Rx VoIP Codec (5) Tx DTMF Mobile 2: (1)VoIP Tx Gain Codec (2) VoIP Rx Gain DTMF -13-...
  • Page 16 LAN, please refer 21.How to setup Asterisk to receive Caller ID from MV-374/MV-378 (page 42) MV-374/MV-378 will send the message as follows in the Packet. From: " caller number " <sip:3001@192.168.0.228>;tag=51088abb User/User(Standard): If you need to register to Asterisk and proxy server, please choose this option.
  • Page 17 User/Tel MV-374/MV-378 will send the message as follows in the Packet. From: " Username " <sip: caller number @192.168.0.228>;tag=7f130947 ※ If you choose this option, please don’t register to Asterisk and proxy server. Please only fill proxy server ip,Username and...
  • Page 18 "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.) -16-...
  • Page 19 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 20 9.4 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 21 Click the serial no,you can view message as follows. -19-...

  • Page 22: Network

    10.Network In Network you can check the Network status, configure the WLAN Settings , LAN Setting and SNTP settings. 10.1 Network Status: You can check the current Network setting in this page. -20-...
  • Page 23 10.2 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 24 10.3 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 25 10.4 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 26 10.5 Slave Settings: Record Slave IP for Master -24-...

  • Page 27: Sip Setting

    11.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. 11.1 In Servcie Domain Function you need to input the account and the related informations in this page, please refer to your ISP Provider.
  • Page 28 (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.
  • Page 29 11.2 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 30 11.3 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. -28-...
  • Page 31 11.4 Codec ID Setting You can setup the Codec ID in this page. -29-...
  • Page 32 11.5 DTMF Setting You can setup the DTMF Setting in this page. -30-...
  • Page 33 11.6 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. -31-...
  • Page 34 11.7 SIP Responses 11.7.1 486(busy here), 503(Service unavailable): When Device are busying, you can select 486 or 505 to response to SIP. 11.7.2 180 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 35 11.7.3 183(Session Progress)-->[It means "on progressing"]: When you turn 183 on, it means you can hear voicemail while GMS side are busying. We recommend you to turn this on if you use SIP Proxy. 11.8 Other Settings Other Settings: you can setup the Hold by RFC and QoS in this page. To change these settings.

  • Page 36: Nat Trans

    12. NAT Transform In NAT Trans. you can setup STUN and uPnP function. These functions can help your VoIP device working properly behind NAT. 12.1 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 37: System Authority

    13.System Authority In System Authority you can change your login name and password. -35-...

  • Page 38: Update

    14.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. 14.1 Update firmware (1) In New Firmware function you can update new firmware via HTTP in this page.
  • Page 39 14.2 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. -37-...

  • Page 40: 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 41: Reboot

    16.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. -39-...

  • Page 42: Ip Setting

    17. 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 43 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 44: 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 -42-...

  • Page 45: Appendix: Setup Mv-374/Mv-378 With Asterisk

    Your mobile <----gsm network----> MV-374/MV-378 <--lan--> Asterisk <--internet--> VOIP provider <--whatever--> landline To do such a call, you just call your MV-374/MV-378 number (it has its own simcard), then you get an invitation tone, then you dial the number which is handled by Asterisk.
  • Page 46 Here are some screen shots showing all the important parameters. You have to note that in all the configuration process, the MV-374/MV-378 is considered as extension '103' of the IPBX. In Bold are the parameters depending on your installation -44-...
  • Page 47 Here the '#' is important to avoid the two stage dialing when you give a call from Asterisk to GSM. The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk. -45-...
  • Page 48 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.
  • Page 49 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 50 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 MV-374/MV-378 is set, you have to configure Asterisk. On that side, you have to setup files as follow : 19.5 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-374/MV-378 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-374/Mv

    20.How to setup Asterisk to receive Caller ID from MV-374/MV-378 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 [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-374/MV-378: address=192.168.66.203:5060; username=1002, displayname=user_1002 test1 pstn call 0928492911(mobile number) MV-374/MV-378 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.
  • Page 54 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> Call-ID: 20fa417265e6a26d0b0aae4f551f06f3@192.168.66.202 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 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...
  • Page 55 0-15 a=sendrecv test 2 SoftPhone call 1002 MV-374/MV-378 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 56 654bf0a2ab0fa9bb118",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 o=1001 5111461 5111501 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 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...
  • Page 57 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 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:...
  • Page 58 --- (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 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>...
  • Page 59 Scheduling destruction of call '7e45b773130f1fc945efcee502f84042@192.168.66.203' in 15000 ms asterisk1*CLI> <-- 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=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...
  • Page 60 User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1002@192.168.66.202> Content-Length: 0 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.66.203:5060: 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...

  • Page 61: Simple Steps

    (1) *,* --->it is two stage dialing. when mobile call in,MV-374/MV-378 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 62 -60-...

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