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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: address=192.168.66.203:5060; username=1002, displayname=user_1002

test1
pstn



call 0928492911(mobile number)



MV-370



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:[email protected]:7331 SIP/2.0
Via: SIP/2.0/UDP 192.168.66.202:5060;branch=z9hG4bK3d0bbaf7;rport
From: "

035678238

" <sip:[email protected]>;tag=as580472a7

-49-

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Summary of Contents for MV-370

Page 1: ...MV 370 VoIP GSM Gateway User Manual PORTech Communications Inc...

Page 2: ...CESSORY ATTACHMENT 3 6 SETTING AND MANAGING VIA WEB PAGE 4 7 SYSTEM INFORMATION 5 8 ROUTE 5 9 MOBILE 10 10 NETWORK 17 11 SIP SETTING 21 12 NAT TRANS 30 13 SYSTEM AUTH 31 14 SAVE CHANGE 32 15 UPDATE 33...

Page 3: ...21 APPENDIX SETUP MV 370 WITH ASTERISK 41 22 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM 47...

Page 4: ......

Page 5: ...all network 2 Functions 2 1 VoIP SIP GSM conversion 2 2 VoIP SIP CDMA conversion 2 3 Voice response for setting and status enquiring Dial in GSM numbers of MV 370 to get voice information or to operat...

Page 6: ...Network cable 3 4 Antenna 3 5 User s Manual When you receive MV 370 package and find it is damaged or incorrect please contact your vendor 4 Dimension and Panel description 2 3 1 3 2 3 4 3 3 14 5cm 3...

Page 7: ...Lan is called light turns off when GSM answered 4 7 LINK Link indicator light green light Light is on when network is connected correctly 5 Accessory attachment 5 1 Connect the network cable both to y...

Page 8: ...The default IP address of MV 370 is http 192 168 0 100 Before accessing the web page please confirm this address is available in your network Enter the default username and password to login Default u...

Page 9: ...ide Please click on the option you would like to set The setting methods are indicated as the following chapters please input the value or select the item according to your situation Note Please remem...

Page 10: ...call 8 1 1 CID caller ID the numbers of incoming call You could set the CID as the following formats 1 The complete number e g 0911111111 2 The prefix part plus e g 0911 This format means any number s...

Page 11: ...tone for the caller to press the IP address proxy extension or any phone number as destination The caller press the IP address on the phone keys 192 168 0 101 as 192 168 0 101 8 1 3 Example of Mobile...

Page 12: ...to LAN settings at the same time Mobile to LAN Speed Dial Settings gets higher priority Mobile to Lan setting will be not available The call is answered with a prompt dial tone for the caller to pres...

Page 13: ...the incoming call You could set the URL as the following formats 1 The complete IP address e g 192 168 0 101 2 The proxy extension numbers e g 103 3 Part of an IP address plus e g 192 168 0 This mean...

Page 14: ...You could dial on your lan phone call any destination number with prefix of 09 When your lan phone and MV 370 had registered and 09 prefix is setted the routing rules at proxy server or Asterisk 4 d...

Page 15: ...uality 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 8 Incomi...

Page 16: ...your need You could leave those default value before you had tried the complete operation of this device 3 LAN Dialtone Gain DTMF Reciver is not good you can adjust gain down 4 ON Off If you use this...

Page 17: ...xy server ip and choose Active on else field empty in sip setting service demain User Tel MV 370 will send the message as follows in the Packet From Username sip caller number 192 168 0 228 tag 7f1309...

Page 18: ...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 this blank to motivate this functi...

Page 19: ...62 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...

Page 20: ...bile SMS Agent 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 a...

Page 21: ...an view message as follows 10 Network In Network you could check the Network status configure the WLAN Settings LAN Settings and SNTP settings 10 1 Network Status Network Status information of current...

Page 22: ...ork Settings Lan Settings You can check the current Network setting in this page The default IP is 192 168 0 100 you could change it to any available IP address or select different IP type to suit you...

Page 23: ...nt network environment to configure the system properly 3 DHCP client you could refer to your current network environment to configure the system properly 4 PPPoE If you have the PPPoE account from yo...

Page 24: ...function to this device Input the primary and secondary IP Address of SNTP Server to get the date time information Also you could set the Time Zone according to your location and set the time to synch...

Page 25: ...Service Domain In this page you should input the data refer to your ISP Maximum is 3 accounts Realm 1 to 3 You could dial out via first SIP account and receive via the three SIP accounts 1 Active clic...

Page 26: ...the Outbound Proxy IP address If your ISP does not provide the information you could skip this item 9 After setting click the Submit button Remember to click Save Charge 10 You can see the Register S...

Page 27: ...can setup the SIP and RTP port number in this page Each ISP provider will have different SIP RTP port setting please refer to the ISP to setup the port number correctly After setting remember to click...

Page 28: ...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 After setting remember to click the Submit button 24...

Page 29: ...11 4 Codec ID Setting You can setup the Codec ID in this page 25...

Page 30: ...11 5 DTMF Setting You can setup the DTMF Setting in this page Note If this device has registered at SIP Proxy Server Asterisk please select 2833 If not please select Inband DTMF 26...

Page 31: ...11 6 RPort Setting You can setup the RPort Enable Disable according to your ISP information After setting remember to click the Submit button 27...

Page 32: ...LE 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 Attempt directly For this function 183 must be turn on 11 7 3 183 Ses...

Page 33: ...n After setting remember to click the Submit button The QoS setting is to set the voice packets priority If you set the value higher than 0 then the voice packets get the higher priority to the Intern...

Page 34: ...N you could setup the STUN Enable Disable and STUN Server IP address This function helps your VoIP device work properly behind NAT Change these settings according to your ISP information After setting...

Page 35: ...13 System Auth In this page System Authority you could change your login name and password 31...

Page 36: ...14 Save Change Please remember this step whenever you submit any setting Click Save Change then Save button the system will restart and make the changed function setting operative 32...

Page 37: ...mware Update Firmware Download the latest firmware then 1 Method select HTTP 2 Code Type select Risc 3 File Location Click the Browse button in the right side of the File Location for the file Please...

Page 38: ...te Default Settings you could restore the factory default settings to the system Click the Restore button then the system returns to default IP http 192 168 0 100 the other settings e g SIP setting ma...

Page 39: ...16 Reboot In this page you could click the Reboot button to restart the system 35...

Page 40: ...P address Default 192 168 0 100 4 Check IP Type 121 IVR announces DHCP is on or off Default off 5 Check Network Mask 123 IVR announces the current network mask Default 255 255 255 0 6 Check Gateway IP...

Page 41: ...mal point 12 Set Gateway IP Address 114xxx xx x xxx xxx Must set Static IP first Enter IP address using numbers on the telephone keypad Use the star key when entering a decimal point 13 Set Primary DN...

Page 42: ...ent 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 VAD CNG AEC LEC...

Page 43: ...MV 370 mobile to lan set route table Step 2 no 2 MV 370 lan to mobile set route table Step 3 Additionally two pcs MV 370 both need to register proxy server Step 4 And proxy server set the route that t...

Page 44: ...ple 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 reque...

Page 45: ...age dialing when lan phone call in MV 370 will provide dial tone and you can enter mobile number 2 specific mobile number when lan phone call in MV 370 will connect with the specific mobile number aut...

Page 46: ...mobile at voip cost 21 2 MV 370 Configuration Once you ve configured everything in the box one good advice is to unplug the power and to restart it By this way you should have all the parameters taken...

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 43...

Page 48: ...ust 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 If you wa...

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 45...

Page 50: ...audio quality 21 4 Asterisk configuration Once the MV 370 is set you have to configure Asterisk On that side you have to setup files as follow 21 5 sip conf GSM VOIP Gateway MV 370 103 type friend us...

Page 51: ...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 sim card mail box thru GSM exten _888 1 SetCallerID xxxxxxx...

Page 52: ...t 1001 qualify yes nat yes host dynamic canreinvite no context internal 1002 type friend secret 1002 qualify yes nat yes host dynamic canreinvite no context internal Add the following setting to etc a...

Page 53: ...168 66 203 5060 username 1002 displayname user_1002 test1 pstn call 0928492911 mobile number MV 370 hear the second dial tone call SoftPhone s number SoftPhone show pstn caller id This Is X Lite rece...

Page 54: ...6 202 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...

Page 55: ...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_100...

Page 56: ...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...

Page 57: ...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 aa From sip 1002 192 168...

Page 58: ...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 UD...

Page 59: ...act 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 algorit...

Page 60: ...n 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 Aster...

Page 61: ...84042 192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Expires 300 Contact sip 1002 192 168 66 203 5060 expires 300 Date Tue 22 May 20...

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