MV-370 / MV-372 VoIP GSM Gateway User Manual PORTech
Contents
1. 32 dll 5 1 2600 5512 32 dll 6 0 2900 5512 Microsoft 80 ma MANIFEST File Common Dialogs DLL Windows NT BASE Client DLL 1 meswvemat d dii dii risycrand dil dded OB 5072r 42 8 0 50727 42 8 0 50727 42 Microsoft C Runtime Library Microsoft C Runtime Library Microsoft C Runtime Library Dy ne SMS SMS na SMS ILE File Configuration Settings USM EXE 478 2 _ tolder SMS shlwapi dll user3z dl File 5 0 2900 5512 5 1 2600 5512 1 187 Shell Light weight Utility Library Windows XP USER Client DLL acuments ws 32 dll 2 1 2800 5512 Windows Socket 2 0 32 Bit DLL zbor s Documents laces Step 6 Please do the configuration as following MV 378 487 gt SMS Notepad J File Edit Format view Total 4 1 192 168 0 100 192 198 0 100 3 182 168 0 100 4 192 168 0 100 374 28 File Edit Format view Help 372 amp 370 29 U File Edit Format View Help Step 7 Run MV SMS program 30 MVsms exe File Edit View Favorites Tools Help 7 s Search Folders i E my_smsiMysms_exelMysms_2. 52 TS SVSleln AUITI Seo 53 16 ne eee een eee Sea eee E 54 T 55 MIRC DOO deed 57 EE tm UU m m UU 58 20 ON RN TR EM E ATE 60 ZO EE 60 20 2 TOPAR HH 60 20552000 _ 60 PRE OIC Cr uli 60 20 5 GSM AM VESTOININ SOT O A md Ot SUN M ND 61 21 Appendix Setup MV 370 MV 372 with Asterisk 61 61 21 2 MV 370 MV 372 Configuration sees 61 zil SKALE A POSON 65 ASICTISK COMMOUFANOM ceed asad 65 22 How to setup Asterisk to receive Caller ID from MV 370 MV 372 67 77 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 2 Function description 2 1 VoIP SIP GSM MV 370 MV 372 conversion 2 2 50 sets of LAN gt MOBILE routes setting gt 50 sets of MOBILE gt LAN routes setting 2 3 Voice response for setting and status dial in from mobile
3. To lt sip 1002 192 168 66 202 gt tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502184042 m 192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Expires 300 Contact lt sip 1002 192 168 66 203 5060 gt expires 300 Date Tue 22 May 2007 03 11 54 GMT Content Length 0 76 23 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 7 gt 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 2 specific extension or IP or phone number when mobile call in MV 370 MV 372 will connect with this specific extension or IP or phone number auto If you want to set specific phone number please note your asterisk need to have route of destination number Lan to Mobile 1 gt is two stage dialing when lan phone call in MV 370 MV 372 will provide dial tone and you can enter mobile number 2 specific mobile number when lan phone call in MV 370 MV 372 will connect with the specific mobile number auto 3 gt It is 1 stage dialing W
4. Setting Setting SIP Respanses Other Settings MAT Transform Update system Authority Save Change Reboot 5 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 To change these settings please following your ISP information When you finished the setting please click the Submit button PORTech STUN Setting Your CTI Partner Route STUN of Mobile 1 S of Mobile 2 Mobile Network STUN Server SIP Settings STUN Port NAT Transform S TUN Setting Update owvster Authority Save Change Reboot 52 On Of On of 4 8 1024 85535 15 System Auth In System Authority you can change your login name and password PORTech Your CTI Partner Route Mobile Network SIP Settings NAT Transform Update Save Change system Authority You could change the login username password in this 53 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 PO Save Change
5. means to add in front for example d2a09 means one stage dialing delete the first 2 codes from your destination number then add 09 in front as the new destination number Example Lan to Mobile 1 MV 370 MV 372 and Lan Phone both need to register proxy server or Asterisk 2 Proxy server asterisk set the route that the prefix of destination number 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 ch 1 MV 370 MV 372 mobile to lan set route table ch 2 MV 370 MV 372 lan to mobile set route table Additionally two channels MV 370 MV 372 both need to register proxy server or Asterisk And proxy server asterisk set the route that the prefix of destination number dial out from ch 2 MV 370 MV 372 The channel 2 MV 370 MV 372 s the first ip 5062 eg http 192 168 0 100 5062 Ree 11 Mobile 11 1 Mobile Status PO Rlech Mobile Status Your CTI Partner 2008 05 16 18 10 Route Mobile 1 Mobile NOSTI Settings Card ID Fwd Settings zii _ SIP Settings Peon PO NAT Transform Incoming IP Name Update Sao PO system Authority Incoming Mob PO Reboot Outgoing Mab 1
6. 192 168 0 100 5050 Fwd to Mobile2 1 192 168 0 100 5062 wd Settings SMS Agent Fwd to Network SIP Settings NAT Transform Update system Authority wave Change Reboot 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 gt Edit gt Advanced gt Accept redirection replies Turn on the Forward Enable therefore the SJ Phone can designate a port which are free to use 18 Profile Options General Initialization Advanced Use short headers Expose software version Use obsolete transfer mechanism m Eestrict caller identity support varies for proxies from different vendors Use standard status messages otherwise messages will be taken from SIP packets Voice mail number or address Remove fancy characters from phone numbers EHI ____ URL Port Fwd to Mobile4 192 168 0 100 5060 Fwd to Mobile2 192 168 0 100 5062 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 f both 5060 port and 5062 port are busying at same time
7. v 0 o root 2737 2737 IN IP4 192 168 66 202 s session C IN 1 4 192 168 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 72 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 z9hG4bK590e92b551233a10a0ae 1944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 9 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 10 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 3ca93a1e response 4d39ccb0dae64 bb2f1341e9896ac1ea 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 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae 1944c1 9b5aa rec 21782 eived 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f84042 1
8. 18 Untitled 1 OpenOffice org Calc File Edit view Insert Format Tools Data Window Help B BAS s BER HOE 81 v 10 4 lt CI se 1 65536 Default Formatting Insert Delete 8 amp Delete Contents Insert Mote 9 Cut Em Copy ES Paste Paste Special Blank B 22 GE dn BA es S oa BF ome 15 222 Default Formattii Formatting Format Cells Insert Delete Delete Contents Insert Bloke C LIE Paste pe Paste Special Step 2 In the Format Cells please select Text Untitled 1 OpenOffice ore Calc File Edit wiew Insert Format Tools Data Window Help amp ef 29 o dh D EN ES arial Cio S 414655356 Format Cells Fraction E Cell Protection Numbers Font Font Effects Alignment Asian Borders Background 1 anit Alig ypography kg 4 Category Format Language 5 Currency Default b Date 7 Time F Scientific 9 10 11 1234 57 Options Decima
9. CODEC Rx E 0 7 SIP From Tel User Standard Answer Delay 0 15 CUD Presentation Suppression 2 Invocation Mobile Code On 1 Code Confirmed LAN Answer Mode Answered Alerted Income Mobile 1 6 Rx uu 1 VoIP Tx Gain 2 2 VoIP Rx Gain 1 VoIP Tx Gain To adjust the volume of LAN side 15 2 VoIP Rx Gain To adjust the volume of Mobile side 3 LAN Dialtone Gain DIMF Reciver is not good you can adjust gain down 4 ON Off If you use this channel please click on Otherwise please click off 9 Routing Range route table 50 sets can share by two channels ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for item 25 49 6 CODEC Tx Gain as above 7 Rx Gain as above 8 SIP From Caller ID transfer el User Standard If you need to register to Asterisk and proxy server please choose this option 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 lt sip 3001 192 168 0 228 gt tag 51088abb e Tel Tel MV 370 MV 372 will send the message as follows in the Packet From caller number sip
10. caller number 192 168 0 228 gt tag 6ac93f7c s Please note lf you choose this option please don t register to Asterisk and proxy server Please only fill and choose Active on else field empty in sip setting service demain e User Tel MV 370 MV 372 will send the message as follows in the Packet From Username sip caller number 192 168 0 228 gt tag 7f130947 s If you choose this option please don t register to Asterisk and 16 proxy server Please only fill proxy server ip Username and choose else field empty in sip setting service demain 9 Presentation CLIR If you need to block the Caller Id for call termination please choose Suppression 10 Mobile PIN you need to 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 13 When you buy Quad band you need to choose your GSM frequency 11 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 PORTech Your CTI Partner Forward Setting p Forward Enable Mobile O Status Fwd to Mobilet
11. 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 You can register three SIP accounts You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from the tree SIP account First you need to click Active to enable the Service Domain then you can input the following items 1 No choose Mobile 1 or Mobile 2 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 6 Domain Server you need to input the Domain Server get from your ISP 7 Proxy Server you need to input the Proxy Server get from your ISP 8 Outbound Proxy you need to input the Outbound Proxy get from your ISP If your ISP does not provide the information then you can skip this item 9 You can see the Register Status in the Status item 10 When you finished the setting please click the Submit button Remember to click Save Charge
12. 40 service Domain Settings Route Mobile 1 Mobile ____ Realm 1 Default Network ACTIVE ON OFF SIP Settings Display Name User 3001 n Port Settings Register Codec Settings Codec IO Setting Register Password Setting Domain Server EO Proxy Serer 61 218 151 230 SIP Responses Other Settings Outbound Proxy NAT Transform status Registered Example Register VoipBuster Realm 1 Default Active off Display Name eny0922 00 User Name jenny0922 Your Voipbuster username Hegister jenny0922 0 00000000 Register Password Your Voipbuster password Domain Server Proxy Server 194 221 62 207 Proxy Server s IP Outbound Proxy Status Reqistered 41 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 PO RTech Ports Setting Your CTI Partner Route Port of Mobile 1 Mobile SIP Port 5060 1024 65535 Network RTP Pott 60000 1024 65535 SIP Settings Port of Mobile 2 Service Damain SIP Port 5062 1024 65535 ET RTP Part 60100 1024 65535 Codec ID Setting DTMF Setting Setting SIP Responses Other Settings NAT Transform Update Syste
13. 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 192 168 66 203 5060 branch 2z9hG4bK672fa67159c222327 5f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 9 192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 12 headers 0 lines Reliably Transmitting NAT to 192 168 66 203 5060 75 OPTIONS sip 1002 192 168 66 203 5060 SIP 2 0 SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK b92dd8a rport From Unknown lt sip Unknown 192 168 66 202 gt tag as5dee3942 To lt sip 1002 192 168 66 203 5060 gt Contact lt sip Unknown 192 168 66 202 gt Call ID 5ebc2211278e2cb 69991 1ad39454d4e 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 192 168 66 203 5060 branch z9hG4bK6 7 2fa6 f59c222327 5f5ee286d2 59 a recel ved 192 168 66 203 rport 5060 From 8 1002 2192 168 66 202 1 4
14. Transform Update system Authority Save Change TCT c c Feboot call will be answered and prompt dial tone again When the caller may enter the 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 eT 10 4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE PORTech LAN To Mobile Table Your CTI Partner Route Page URE Call Num Select it Mobile To Lan Speed Dial E Lan To Mobile Settings Mobile SIP Settings NAT Transform _Update system Authority save Change Reboot The 370 372 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 192 168 0 101 or proxy server s extension If a simple is entered means no restriction for the incoming IP address Call Num 1 may enter the whole number e g 0911111111 2 a simple 2 stages dialing The call will be answered and prompt dial tone again to receive the called number as the 12 destination e g 0911111111 or 0911111111 3 Z d n a ppp for one stage dialing Is option d n means to delete the beginning n codes
15. have all the parameters taken into account To have the MV 370 MV 372 to work with Asterisk you need first to 61 configure the box 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 WAN Settings You could configure the VAM settings in this page WAM Setting IP Co Fixed IP DHEP Client PPPoE IP P370 IP Mask 255 255 255 0 sateway Router DNS Servert 158 95 192 1 B DNS Server 168 95 1 1 PPPoE Setting User Mame Password Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM LAWN To Mobile Table Page 1 II Select 0 Your Asterisk IF Ht 1 E 5 62 Mobile To LAN Table Select Authorised Mobile 103 Another Authorised Mobile 103 2 3 4 5 The mobile number you give 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
16. telephone key pad Use the star key when entering a decimal point 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 otatic IP DHCP PPPoE 20 3 Codec G 711 u Law G 711 a Law G 723 1 5 3k 723 1 6 3k G 29A G 29A B 20 4 Voice Quality VAD 60 CNG AEC LEC Packet loss 20 5 GSM MV 370 MV 372 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 21 Appendix Setup MV 370 MV 372 with Asterisk 21 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 gt MV 370 MV 372 lt lan gt Asterisk lt internet gt 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 If you have some special deals with your mobile operator like free special number you can call your MV 370 MV 372 for free You can then call all around the world from your mobile at voip cost 21 2 MV 370 MV 372 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
17. 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 If you want to allow any number just set in that field but beware of the bill 63 Service Domain Settings Realm 1 Default Active ON OFF Display Name LR User Mop Register Mame 103 L Register Password Domain Server Asterisk Proxy Server status Mot Registered Outbound Proxy Once Asterisk configuration is made you should get Registered on the Realm1 Codec Settings Codec Priority Codec Priority 1 6 711 U lawy Codec Priority 2 5 711 a lav Codec Priority 3 Not Used Codec Priority 4 Mot sed Codec Priority 2 Mot Llsed Codec Priority Mot Used Codec Priority 7 Not Used Codec Priority Mot sed RTP Packet Length 5711 amp G 228 20 ms 3 7 23 ms 5 123 5 3K 22 E SE C On of VAD AD On of 64 It is very important to use only u law or a law as all DT MEF is inband So if you want to be abl
18. 145 5060 username 1000 displayname user_ 1000 X Lite lt 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 68 v 192 168 66 203 Live S h BRO KAO HAO BARZU NAD 4 wp ae VoIP Web Management gt 9 Mobile Votp Service Domain Settings You could set information of service domains in this page Route gt Mobile 4 Mobile 1 34 Network Active On C Off SIP Settings gt Display Name user 1002 8 NAT Trans User Name Register Name System Auth Register Password ess Save y ERA Domain Server 9216866200 1 Proxy Server 193246866202 0 Update gt Outbound Proxy 19246866202 Reboot Status Registered Active C Of Display Name User Nam S Register Nam Register Password E oe eee 5109 4 test1 pstn gt call 0928492911 mobile number gt MV 370 MV 372 gt hear the second dial tone call SoftPhone s number gt SoftPhone gt 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 SIP 2 0 UDP 192 168 66 202 5060 branch z9h G4b K 3dObbaf rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a To lt sip 1001 192 168 66 145 7
19. 2 4 Series connections to save bills 2 5 Standard SIP RFC2543 RFC3261 protocol gt Communicates with other gateway or PC 3 Parts list Please check the parts for any missing parts If do please contact our agents 3 1 MV 370 MV 372 main body 3 2 Power adaptor AC DC 110V AC 12V or 220V AC 12V DC 3 3 Network cable 3 4 Antenna 3 5 User Manual 3 1 MV 370 3 1 MV 372 3 4 5 MV 370 Panel description 145 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 5 5 MOBILE GSM indicator light green light Light flashes when GSM status is normal light turns on constantly when GSM is called 5 6 LAN LAN indicator light green light Light flashes when Lan is called light turns off when GSM answered 5 7 LINK Link indicator light green light Light is on when network is connected correctly 6 MV 372 Panel description 52 V 372 Mobile VolP 6 1 E 2 he Your Partner POR 6 2 6 3 6 4 6 5 6 6 6 76 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 Powe
20. 331 gt Contact lt sip 1002 192 168 66 202 gt Call ID 20fa417265e6a26d0b0aae41551f06f3 9 192 168 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 69 Content Type application sdp Content Length 242 v 0 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 z9h G4b K 3dObbaf rport From 035678238 lt sip 1002 192 168 66 202 gt tag as5804 72a To lt sip 1001 192 168 66 145 7331 gt tag 677373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae41551f06f3 9 192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x Content Length 254 v 0 07 1001 4804366 4807851 192 168 66 145 s X Lite C IN 1 4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101 a rtpmap 0 pcmu 8000 70 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 gt call 1002 gt MV 370 MV 372 gt hear second dial tone and call pstn gt pstn answer gt show caller id mobile number 092849291 1 This Is X Lite receiving packet T
21. 9 17 48 17 11 idle 28 17 48 23 182 158 0 82 1388 gt CH 11 0129 1 48 43 CH 11 idle 104 15 06 48 49 Status Dial Peer 29 2008 14 54 07 Sel Log Status Set remote state 192 1656 0 100 5060 amp j 192 168 0 100 5082 192 158 0 100 5064 192 165 U 1 00 5066 192 165 0 1 00 5068 192 165 0 1 00 3n 192 168 0100 5072 132 165 U 1 00 5074 182 158 0 110 5060 192 169 0 110 5062 192 169 0 110 5064 ma ny des 182 158 0 11U 5066 192 58 0 110 5068 182 158 0 110 5070 192 165 0 110 5072 182 158 0 11U 5074 081104 15 07 14 50 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 priority If you set the value higher than O then the voice packets will get the higher priority to the Internet But the QoS function still need to cooperate with the others Internet devices PO Rlech Other Settings Your CTI Partner Rout Hold by RFC of Mobile 1 of Hold by RFC of Mobile 2 On Off Mobile Network Voice QoS 40 0 63 SIP Settings SIP Ges 40 0 63 Service Domain SIP Expire Time 300 0 86400 sec Port Settings Codec Settings Codec IL Setting
22. 92 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 Transmitting NAT to 192 168 66 203 5060 2 0 401 Unauthorized Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae 1944c1 9b5aa rec eived 192 168 66 203 rport 5060 From sip 1002 192 168 66 202 tag 4e36de8f1 To lt sip 1002 192 168 66 202 gt tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502184042 m 192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY WWNW Authenticate Digest algorithm MD5 realm asterisk nonce 5def9231 Content Length 0 Scheduling destruction of call Te45b773130f1fc945efcee502184042 m 192 168 66 203 in 15000 ms asterisk1 CLI gt lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 4 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK6 7 2fa6 7f59c22232 5f5ee286d2 597a From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 9 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 11 REGISTER Expires 300 Authorization Digest username z 1002 realm asterisk nonce 5def9231 response 046a412f4e7ed4 e98fd507416994a80a uri sip 192 168 66 202 algorithm MD5 User Agent CMI CM5K Content
23. Connect the equipment into an outlet on a circuit different from that to which the receiver 1s connected Consult the dealer or an experienced radio TV technician for help Operation 15 subject to the following two conditions 1 this device may not cause interference and 2 this device must accept any interference including interference that may cause undesired operation of the device 78 FCC RF Radiation Exposure Statement 1 This Transmitter must not be co located or operating 11 conjunction with any other antenna or transmitter 2 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 19
24. LT SETTINGS WILL BE LOST This will include network and service provider data 3 Check IP Address 120 IVR will announce the current Default 192 168 0 100 4 Check IP Type 121 IVR will announce if DHCP in default OFF 5 Check Network 1234 IVR will announce the current 255 255 255 0 Check Gateway 124 IVR will announce the current IP gateway IP address Address IE Default 192 168 0 254 58 0 3 Check Primary DNS Server 125 Version Set 111 client Set Static IP Address 1 12XXX XXX XXX IVR will announce the current setting in the Primary DNS field Default 192 168 0 1 IVR will announce the version of the firmware running The system will change to DHCP Client type DHCP will be disabled and system will change to the otatic IP type Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point Set Network 1 13 Must set Static IP first oet Gateway IP Address oet Primary DNS oerver 11 115 XXX 59 Enter value using numbers the telephone key pad Use the star key when entering a decimal point Must set Static IP first Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point Must set Static IP first Enter IP address using numbers on the
25. MS Complete 33 MY SHS 371 tt HelpiH send SMS Complete 1 0935386862 2009 2725 09 59 36 2 0935386862 2009 2725 09 59 26 4 0931266207 2009 2 25 09 59 27 3 0912062361 2009 2725 09 59 27 1 0935386862 2009 2725 09 59 13 2 0912062361 2009 2725 09 59 05 4 0931266207 2009 2725 09 59 05 3 0881085825 2009 2725 09 59 05 Login Telnet system ohio Message Total 34 11 5 use AT Command via Telnet or your program Allows your program or Telnet Send receive SMS with AT Command Port 23 username password user level 1 Please enter account and password command logout module modulel module modulel 2modulel choose module got press ctrl to release module 1 Enter 1 you can see ate1 your at command below at cmgf 1 atom s 9911129656 Enter atsomgs phone number gt test gt 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 Network Status Your CTI Partner Ethernet WAN Interface LAN Interface Mobile Type Fixed IP Client Fixed IP Client IP 192 168 0 122 142 163 5 102 Network Mask 209 295 255 0 255 255 255 0 Gateway 182 158 0 254 192 158 0254 AAN me
26. MV 370 MV 372 VoIP GSM Gateway User Manual MV 3 2 Content teet teet 1 2 F nction descriIpUOn xo x ee eee 1 1 4 Dimension 14 5cm x 17cm X3 a 2 2 MV 370 Panel description sas tuf v ie ohne nem sete 3 5 MV 35 2 Panel d scrIDUOD ccn ti rh tr 4 Te 5 o 6 9 System Information Jadre rodeo nn 7 TU SOUPE NE 7 10 1 Mobile EOD cue Tm 8 19 2 Call Back el VICE 50 86165 10 10 3 Mobile to LAN Speed Dial Settings 11 tO SS ee ee 12 jsp 14 r 14 TEZ MODI SEUN cl ci titi n 15 11 3 Mobile Forward 0 17 114 Mobile SMS Agel a eta aevo duet eta 19 11 5 Send Bulk of SMS via Microsoft Excel 22 Fei rolg er E 36 40 13 7 1 486 busy here 503 Service 47 19 7 2 190 FRING HON OM zl nn lin 47 TOS SESSION 47 EE 47
27. 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 9 Incoming IP The IP address of the last incoming call from LAN 7 JOutgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call from MOBILE 6 Incoming Name proxy server name 9 Outgoing Mob The called number of the last outgoing call to MOBILE 14 11 2 Mobile Setting Only change mobile into on or just click Mobile Setting submit no need to click Your CTI Partner save change _ Route 1 Tx Gain 0 12 GIP Rx Gain 11 0 15 Mobile 3 LAN Gain 9 0 12 status Settings 4 Mobile1 ON OFF Fwd Settings 5 Routing Range 0 to 49 01 497 6 CODEC Tx e 0 7 7 CODEC Rx Gai 6 mer Network 8 SIP From Tel User Standard Answer Delay lo 0 15 12 SIP Settings 9 CLID Presentation Suppression Invocation NAT Transform Mobile PIN Code 4 _ 10 U Code Confirmed Update 1 LAN Answer Mode 2 Answered Alerted Income system Authority Feboot Routing Range to 0 40 CODEC Tx 6 0 7
28. Proxy server or Asterisk Phone number SIP Proxy Server or Asterisk need to set the route of this phone number 10 2 Call Back Service 50 sets PO RTech Your CTI Partner Mobile To LAN Table Route Page 1 CID URL Select 0933579613 it 566933579613 H Lan To Mobile Settings Mobile Network SIP Settings _NAT Transform Update _ system Authority ux Oo 4 c Ow b CJ Change Reboot Delete Selected Delete All Add New Position D 48 Ex 0911111111 0911 URL 192 188 0 1 251 Add reset You can set call back service as the following steps 1 set the phone number here up to 50 sets 2 URL is the command of call back 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 370 MV 372 and prompt a dial tone 10 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 PORTech Mobile To LAN Speed Dial Your CTI Partner Route Name NN ET TT 192 158 0 107 LI abile an Mobile To Lan Speed Dial an To Mobile Settings Mobile Network SIP Settings NAT
29. You can use mouse to click the function you want to set up PORTech Mobile VoIP2 Route Model Type hea Mobile Model Description oh S001 S00MHz TC35 Network Firmware version Tue 4 14 15 35 2006 Codec Version Jul 24 10 55 05 2006 SIP Settings O no d Marth Raad Taichung Taiwan NAT Transform T l 986 4 23058000 Update Fax OOb 4 2302 7585 System Authority E Mail salesicipartech cam tw Wah Site http www portech com tw Save Change ie 2008 PORTech Communications Inc 10 Route Important The route table 50 sets can share by two channels The setting please refer 11 2 Mobile setting ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for ttem 25 49 10 1 Mobile TO LAN Settings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN PORTech Mobile To LAN Table Your CTI Partner Route Page EC Mobile Speed Dial O Lan To Mobile Settings Mobile Network SIP Settings _ Transform Update System Authority save Change Reboot M Delete Selected Delete All Add New Position 0 49 0911111111 09117 URL 192 168 0 1 25 The MV 370 MV 372 will transfer to the URL according to the caller ID of the Mobile 1 may
30. e to do some DISA when you call from GSM to Asterisk it has to be one of these 2 codecs Mobile Setting VoIP Tx Gain 0 12 VoIP Rx Gain _ 0 415 LAM Gain Routing Range to 49 0 49 CODEC Tx 6 2 CODEC Rx From Tel User Standard Answer Delay 0 15 CLID Presentation Suppression Invocation These settings seem to be ok just adjust 21 3 Antenna position Another important thing is to properly place the provided antenna If your gsm reception is good you should get around 18 or 19 as Signal Quality in the Mobile Status page With that level of signal quality your audio quality will be very good 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 GSM VOIP Gateway MV 370 MV 372 103 type friend 65 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 lt 103 gt host dynamic nat no Gateway is not behind a NAT router canreinvite no Typically set to NO if behind NAT ins
31. ecure very qualify yes disallow all allow ulaw prefered codec for DIMF detection allow alaw 21 6 extensions conf HEFT GSM Gateway incoming calls gateway exten gt 103 1 exten gt 103 2 DigitTimeout 3 give enough time to do second stage dialing exten gt 103 3 ResponseTimeout 5 exten gt 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 gt _ 888 1 SetCaller D xxxxxxxxxx exten gt 888 2 Dial SIP EXTEN 103 60 r exten gt 888 3 66 22 How to setup Asterisk to receive Caller ID from MV 370 MV 372 trixbox 2 2 SIP Softphone e SJPhone 1 60 289a e X Lite 1105x 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 type friend _67 secret 1002 qualify yes nat yes host dynamic canreinvite no context internal Add the following setting to etc asterisk extensions conf internal exten gt 1000 1 Dial SIP 1000 exten gt 1001 1 Dial SIP 1001 exten gt 1002 1 Dial SIP 1002 configure trixbox 2 2 address 192 168 66 202 5060 SJPhone address 192 168 66
32. 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 The item which has more digits will have higher priority If the digits are the same then former one gets the higher priority 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 it means refuse to transfer 31 entered it means 2 stages dialing The call will be answered and prompt dial tone again to receive the IP address sip extension or any phone number as the destination The caller may enter the IP such as 192 168 0 1015 If the device have register proxy server Asterisk you can enter any destination phone number Please note the proxy server Asterisk need to set the route of destination phone number Example 1 Mobile to Lan 0932 0911123456 MV 370 MV 372 have register proxy server Asterisk The proxy server Asterisk have the route 09 When the callers 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
33. est 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 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt Contact lt sip 1001 192 168 66 145 7331 gt Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE Proxy Authorization Digest username 1001 realm asterisk nonce 0d3b28 9 response 8aaaaadbd5ad53 654bf0a2ab0fa9bb1 18 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 v 0 021001 5111461 5111501 192 168 66 145 s X Lite 7 c IN 1 4 192 168 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 331 From user_1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 lt sip 1002 192 168 66 202 gt tag as2a2fbt98 Call ID F 4B32CA6 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 lt sip 1002 192 168 66 202 gt Content Type application sdp Content Length 242
34. exe comdlgs dll A 6 0 2900 5512 Common Dialogs DLL 5 msvcm d dii dded c B D S0 727 42 Microsoft C Runtime Library ns Vsms exe 2 Mysms exe older 5 5 File 1 187 KB 3cuments vys 32 4 stor s Documents A 5 1 2600 5512 Windows Socket 2 0 32 DLL ech laces Step 8 1 Open File Taol T Open FiletF kernel3z dll 5 1 2500 5512 Windows Client DLL mswvepatid dll 8 0 50727 42 Microsoft C Runtime Library 5 5 ILE File 478 shlwapi dil 6 0 29600 5512 Shell Light weight Utility Library LIBI y LE wire A 31 Microsoft 80 MANIFEST File 1 msvcr amp d all o 0 50 27 42 Microsoft C Runtime Library MV SMS Configuration Settings 1 user3z dil 5 1 2600 5512 Windows USER client DEL sli ibr 2 Open the Excel file that you just saved T PM MV_SMS TookT Documents E My Computer MyRecent 4 Network Places Documents REG Desktop 2 My Documents ga Computer My Network File name Places Files of type Step 9 Sending 32 7 m MY SMS 32 txt Tooli Login Telnet system Start System Waiting Step 10 Send S
35. hen lan phone and MV 370 MV 372 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 MV 370 MV 372 All changes both need to click save and change 77 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 These limits are designed to provide reasonable protection against harmful interference in a residential installation This equipment generates uses and can radiate radio frequency energy and if not installed and used accordance with the instructions may cause harmful interference to radio communications However there 1s no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or television reception which can be determined by turning the equipment off and on the user 1s encouraged to try to correct the interference by one or more of the following measures Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver
36. l places o Negative numbers red Leading Thousands separator Format code E 44 Please do this action for Blank A and both 253 Step 3 Blank A 15 for you to key phone numbers Blank 15 for you to key text File Edit View Insert Format Tools Data Window Help D C E Ss 0 een d How Are You 098888888 Step 4 Save the file 24 dit View Insert Format Tools Mew E A Open Recent Documents JI NN a B H Data Window Help 3 Wizards k EXCESS Are You Export Export as PDF Send kr Properties Digital Signatures Templates Save the type as Unicode Text 225 1 Tools M Documents 49 Computer Recent Network Places Documents My Documents E Computer File name test Network Places Save as type Unicode Text A Step 5 Open MVsms exe gt MV SMS Configuration Settings 26 File Edit View Favorites Tools Help Back ag 2 Search le Folders Addres E Go Folders
37. le channel to dial out E g When the first port is busy MV 378 will use the second port to dail out and so forth _47 Edit DialPeer ini Window 14 5070 Xpos 512 15 5072 Ypos 252 16 5074 Width 47 1 RtpPort Height 399 1 60000 Info 2 60002 Total 16 3 60004 VoipIP 4 60006 1 192 168 0 100 5 60008 2 192 168 0 100 6 60010 3 192 168 0 100 7 60012 4 192 168 0 100 8 260014 5 192 168 0 100 o SEES loy 9260000 6 192 168 0 100 10 60002 7 192 168 0 100 11 60004 8 192 168 0 100 12 60006 9 192 168 0 110 13 60008 10 192 168 0 110 14 60010 11 192 168 0 110 15 60012 12 192 168 0 110 16 60014 13 192 168 0 110 PtcPort 14 192 168 0 110 1 40000 15 192 168 0 110 2 40000 16 192 168 0 11 3 40008 SipPort 4 40008 1 5060 5 40016 2 5062 6 40016 3 5064 7 40024 4 5066 8 40024 5 5068 9 40000 6 5070 10 40000 7 5072 11 40008 8 5074 12 40008 9 5060 gt 13 40016 10 5062 14 40016 11 5064 15 40024 12 5066 16 40024 13 5068 48 Dial Peer Oct 29 2008 14 54 07 File Help Status Set DateTime event 0729 15 11 08 Start SIP Server 192 168 0 3 5060 0 29 15 11 08 STUN server port 3478 3479 0129 15 13 14 CH 11 idle O49 15 15 02 192 158 0 91 5050 CH 11 0810129 15 15 53 CH 11 idle 0129 17 23 45 CH 11 idle 029 17 47 22 182 158 0 82 1388 gt 11 uz 1 47 54 CH 11 idle 029 17 47 55 192 158 0 92 1398 gt CH 11 0841012
38. m Authority save Change Reboot 42 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 PO Rlech CTI Codec Settings Route Mobile Codec Priority Codec Priority 1 711 Network Codec Priority 2 6 711 law SIP Settings Codec Priority 3 23 Codec Priority 4 6 729 v service Port Settings Codec Priority 5 G 725 16 Codec Settings Codec Priority B 5728 24 Codec ID Setting mS DTMF Setting Codec Priority 5 726 32 RPort Setting Codec Priority 8 2 726 40 5 SIP Responses Other Settings RTP Packet Length NAT Transform Update G 723 30 ms 571 15725 20 ms system Author Save Change _____ 6 723 5 3K 5 723 5 4K On of Reboot Voice VAD Voice VAD On off 43 13 4 Codec ID Setting You can setup the Codec ID in this page PORTech Codec ID Setting Your CTI Partner You could set the
39. o 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 PORTech Your CTI Partner Status WAN Settings Settings Settings NAT Transform Update _System Authority Save Change Reboot LAN Settings LAN Setting IP Mask Tz 165 0 102 555 255 255 0 0003720089989 DHCP Server DHCP Server start IP End IP ease Time 36 Goff 150 200 l U dd hh 12 4 SNTP Settings SNIP 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 When you finished the setting please click the Submit button SNTP Settings Your CTI Partner Y ou could set the SMTP servers in this Route Mobile SMTP 0208 Network status WAN Settings secondary server 208 184 49 9 LAN Settings Settings Time Zone GMT hh mm SIP Settings sync Time D dd hh mm MAT Transform Update System Authority Save Change Reboot 39
40. r 000357 009999 0003 008888 12 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 If you have the PPPoE account from your Service Provider please input the Username and the Password correctly 3 The Bridge Item is to setuo the system Bridge mode Enable Disable If you set the Bridge On then the two Fast Ethernet ports will be transparent 4 When you finished the setting please click the Submit button 36 PORTech Your CTI Partner Route Mobile Network WAM Settings LAN Settings SMTP Settings SIP Settings NAT Transform Update System Authority Change _ Reboot WAN Settings Y ou could configure the WAN settings in this page Network Mode Bridge WAN Setting IF Type Fixed IPF ODHCP Client PPPoE IP 192 168 0 122 Mask 255 255 255 0 Gateway 192 168 0 254 DNS Server 168 95 192 1 DNS Sener2 168 95 1 1 00037 009999 PPPoE Setting 12 3 LAN Settings You can check the current Network setting in this page 1 The TCP IP Configuration is to setup the WAN 8 network environment You may refer t
41. r LED Light up when power is normal 6 6 VoIP1 an indicator light of VoIP1 6 7 2 an indicator light of VoIP2 6 8 LINK Indicator Light up when network is connected 7 CABLING 7 1 Connect the internet cable from HUB to the WAN connector of the MV 372 If you need to stack up more MV 372 you can stack up as follows How to stack up 7 2 Connect the antenna and put it in proper position to get the best Signal reception 7 3 Insert the SIM card from back of the main body take the slide off first 7 4 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 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 Login POR Tech VolP Enter vour username and password to login VoIP server L sernarme Password Eg Remember last login Enter the username and password for authentication default username voip password 1234 The page follows when the username and password are correct 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
42. s Codec Settings Codec ID Setting Setting Setting SIP Responses Other Settings NAT Transform Update system Authority save mamie Reboot 46 13 7 SIP Responses PORTech SIP Responses Setting Route _ Mobile 456 Busy here Network C 503 Service Unavailable Service Domain OM OFF 180 Ringing Auto force ta ON if 183 was OFF Settings OON OFF 183 Session Progress Codec Settings BIBLE Setting OON OFF Download Dial Peer Other Settings Dial Peer Configuration Table corresponding IP NAT Transform please read next page Update System Authority If you have dial peer server Sip Save Change server Asterisk set GSM route please set Dial menani Peer 5 IP 13 7 1 486 busy 903 Service unavailable When Device is busy 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 prompt voice directly For this function 183 must be turn on 13 7 3 183 Session Progress It means progressing When you turn 183 on it means you can hear the prompt voice while GSM side is busy We recommend you to turn this on if you use SIP Proxy 13 7 4 Dial Peer Lan to mobile Dial peer software will look for availab
43. s You have to save changes to effect them Route Mobile cave Changes Network SIP Settings NAT Transform Update Mew Firmware Default Settings System Authority Save Change Reboot 54 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 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 4 Select the correct file you want to download to the system then click the Update button 5 Please click update default setting after update firmware PORTech Update Firmware You could update the newest firmware PCB mark 261236 Route Mobile Method 9 HTTP TFTP Network SIP Settings Transform Update File Lacatian i Default Settings TFTP Server 192 159 1 250 System Authority Save Change _ Reboot 55 17 2 Restore Default Settings In this page Update Default Settings you could restore the factory default settings to the system All set
44. ting will restore default setting IP will retain original IP as usual not default IP Po Rlech i Restore Default Settings Your CTI Partner Y au could click the restore button to restore the factory settings Route Restore default settings Network SIP Settings NAT Transform Update Mew Firrmvvare Default Settings system Authority Save Change Reboot 56 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 PORTech Reboot System Your CTI Partner You could press the reboot button to restart the system Route Mobile Reboot system Network SIP Settings NAT Transform Update System Authority Change A 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 VolP GSM Gateway enters the IP setting mode The operator may dial in the mobile number during this period to set or query the network parameters tem 1 1958 After you hear Option Successful hang up Unit will reboot automatically 2 Reset 1988 All setting include IP both restore to default setting WARNING ALL User Changeable NONDEFAU
45. value of Codec ID in this page Route Mobile Codec Type ID Default Value 5728 16 0 23 85 255 23 276 24 ID 22 v SIP Settings anes 2 6726 32 ID 95 255 2 G 26 40 ID 21 95 255 21 2833 ID 101 service Domain Settings Codec Settings Codec ILI Setting etting Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 95 255 101 44 13 5 Setting You can setup the DTMF Setting in this page PO RTech Your CTI Partner Route Mobile Network SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting FR Part Setting SIP Responses Other Settings NAT Transform Update System Authority save Change Reboot DTMF Setting Mobile DIME Transfer to Lan 2833 C Inband Send SIP Info Mobile DTMF debounce range 40 200 default all step 1Ums 45 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 RIech RPort Setting Your CTI Partner Route of Mobile 1 On Off Mobile FP ort of Mobile 2 On Off Network SIP Settings Service Domain Port Setting
46. you can set up Fwd to External then you can transfer the phone call to another designate device 11 4 Mobile SMS Agent 19 Read received SMS sMs Agent Your CTI Partner Route _ Mobile 1 Ready Ill Rx List Mobile Mobile 2 Mot Ready Ill Rx List Status Settings Fwd Settings SMS Sender Encode ASC ASCII zbit w Network Ma Dest Num SIP Settings NAT Transform Update Message system Authority Save Change You have 160 ASC chars remaining for your description Reboot 1 Rx List Read received SMS 2 Dest Num the phone number 3 Message Please fill the message that want to send to receiver 2 mode ASC7 ASCII 7 bit UCS2 Unicode 16 Maximum Number of ASC chars f cend Now When you click Rx List you can view all received SMS as follows SMS Rx List Read Status RemotelD 1 REC READ 9969356114545 09 01 01 19 34 22 2 REC READ dob 535606 08 03 12 16 25 27 Click the serial no you can view message as follows 20 SMS Reader i 99599539696 08 03 12 15 25 27 Serial can send SMS and receive SHS Back Delete ela 11 5 Send Bulk of SMS via Microsoft Excel First of all please open a new Excel file Step 1 Format Cells Here we need you to format cells to Text first Please click mouse right key and choose Format Cells Blank A
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