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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

    16.Reboot ........................ 43 17. IP Setting ......................44 18.Specification ...................... 46 19. Appendix: Setup MV-374/MV-378 with Asterisk..........47 20.How to setup Asterisk to receive Caller ID from MV-374/MV-378....54 21. Simple Steps ..................... 63 www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • 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) MV-378 (3.1) MV-374 (3.2) MV-378 (3.2) MV-374 (3.4) (3.3) (3.5) option 4.Dimension : 30x28x4 cm www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • 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: Mobile To Lan Settings

    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: Call Back Service (50 Sets) **New Feature

    Application: a. Call MV-374/MV-378 b. MV-374/MV-378 will detect the phone number is in call back list or not c. If yes, MV-374/MV-378 will reject the call, and call it back d. You will receive the call from MV-374/MV-378, and prompt a dial tone...

  • Page 11: Mobile To Lan Speed Dial Settings

    8.3 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time,MV-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 12: Lan To Mobile Settings

    8.4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE. The MV-374/MV-378 will transfer to the mobile number according to the incoming URL *URL：The IP address of the incoming call.
  • Page 13 (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 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 Only change “mobile” into “on” or “off”,just click “submit”, no need to click “save change” (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- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • 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: Mobile / Forward Setting

    9.3 Mobile / Forward Setting : When the first route are busying, SIP can transfer phone call to another free route. When the device are busying, the phone call can be transfer to another device (external equipments). "Forward Enable" is not motivate on Defualt value. So please, mark "Forward Enable"...
  • 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: Mobile / Sms Agent

    9.4 Mobile / SMS Agent : Read received SMS 2 mode: ASC7(ASCII 7 bit) UCS2(Unicode 16 bit) (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.
  • Page 21 Click the serial no,you can view message as follows. -19- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 22: Use At Command Via Telnet Or Your Program

    9.5 use AT Command via Telnet or your program Allows your program or Telnet Send/receive SMS with AT Command available in PCB194A (approximately after April , 2008) Telnet PORT Corresponding port as follows: Master ip: 23 SLAVE 1 :8023 SLAVE 1 :8123 SLAVE 1 :8223 Please enter account and password...

  • Page 23: 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. -21- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • Page 24 10.2 WAN Settings: WAN IP (Master) Default: 192.168.0.100 Slaver1 : Master ip:8080 Slaver2 : Master ip:8180 Slaver3: Master ip:8280 WAN IP Corresponding port 5060 5062 5064 5066 5068 5070 5072 5074 (1) The TCP/IP Configuration item is to setup the WAN port’s network environment.
  • Page 25 10.3 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 10.4 Slave Settings: Record Slave IP for Master **PLEASE don’t change this page** Important!!! -24- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • 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 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- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 30: Codec Id Setting

    11.3 Codec ID Setting You can setup the Codec ID in this page. -28- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 31: Dtmf Setting

    11.4 DTMF Setting You can setup the DTMF Setting in this page. -29- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • Page 32 11.5 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- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 33: 486(Busy Here), 503(Service Unavailable)

    11.6 SIP Responses Dial Peer Configuration Table corresponding IP (please read next page) *** If you have dial peer server, Sip server/Asterisk set GSM route,please set Dial Peer server’s IP** 11.6.1 486(busy here), 503(Service unavailable): When Device is busy, you can select 486 or 505 to response to SIP. 11.6.2 180 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...
  • Page 34 Edit DialPeer.ini [Window] 14=5070 Xpos=512 15=5072 The second Ypos=252 16=5074 MV-378 Width=471 [RtpPort] Total ip / port Height=399 1=60000 [Info] 2=60002 Total=16 3=60004 [VoipIP] 4=60006 1=192.168.0.100 The first 5=60008 2=192.168.0.100 6=60010 MV-378 3=192.168.0.100 7=60012 The first 4=192.168.0.100 8=60014 MV-378 5=192.168.0.100 9=60000 6=192.168.0.100 10=60002...
  • Page 35 -33- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • Page 36 Status The first MV-378 doesn’t register dial peer software The 2,3ch of Second MV-378 idle The 1,4-8ch of Second MV-378 turn off -34- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • Page 37 11.7 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 38: Nat Transform

    ISP information. When you finished the setting, please click the Submit button. If you want to set up NAT for MV-374/MV-378, you should install STUN Server first. (Or it can only allow one-way call) The initial setting of STUN Server is ON You can download STUN Server here(Free): www.myvoipapp.com...

  • Page 39: System Authority

    If you set MV-374/MV-378 at Private IP, you should use STUN Server's private IP If you have installed Dial Peer server,please fill Dial Peer server’s ip directly 13.System Authority In System Authority you can change your login name and password.

  • Page 40: 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 MV-374 have to update 2 times MV-378 have to update 4 times Master IP : 8280 (first update better)
  • Page 41 Slaver2 192.168.0.100:8180 Slaver3 192.168.0.100:8280 -39- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • Page 42 (1) In New Firmware 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 the File Location or you can type the correct path and the filename in File Location blank.
  • Page 43 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. Factory all: all setting include ip will restore default setting. -41- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 44: 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 45: 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. -43- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 46: 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 47 IVR will announce the current Check Primary #125# 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...

  • Page 48: 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 -46- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 49: 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 50 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 -48-...
  • Page 51 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. -49- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...
  • Page 52 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 53 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 54 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 55 => _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() -53- www.InternetVoipPhone.co.uk | sales@internetvoipphone.co.uk | 0800 088 4846...

  • Page 56: How To Setup Asterisk To Receive Caller Id From Mv-374/Mv-378

    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 57 [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 58 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 59 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 60 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 61 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 62 --- (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 63 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 64 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 65: 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 66 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 67 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-378

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