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MV-370 VoIP GSM Gateway User Manual PORTech
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1. If not please select Inband DTMF 26 11 6 RPort Setting You can setup the RPort Enable Disable according to your ISP information After setting remember to click the Submit button PORTech npo Setting Your CTI Partner EOM RPart of Mobile On O Off Mobile Network SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 27 11 7 SIP Setting SIP Responses PORTech Your CTI Partner SIP Responses Setting Route Mobile 9 486 Busy here Network 503 Service unavailable SIP Settings E SIP Responses Service Damain OON OOFF 180 Ringing Auto force to ON if 183 was OFF Port Settings OON OFF 183 Session Progress Codec Settings Codec ID Setting DTMF Setting EPaort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 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 Attempt directly For this function 183 must be turn on 11 7 3 183 Session Progress gt It means on progressing When you turn 183 on it means you can hear Voice Attempt while GMS side
2. 1000 type friend secret 1000 qualifyzyes 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 secret 1002 47 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 66 145 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 48 VoIP Web Management Windows Intemet Explorer iwi G E htp 192 168 66 203 0gincgi zl ssx 6 Seam y Plz BRO SEO HAO PRO TAD HAD mee Sp ae ae abala VE Se VolP Web Management m dig Mobile Voi Service Domain Settings You could set information of service domains in this page Route gt Mobile No Mobile 1 kat Realm 1 Default Active On Of SIP Settings gt Display Name fuser_1002 User Ni 1002 NAT Trans gt iain Register Name 1002 System Auth Register Password Imm Save Change Domain Server 192 168 66 202 Proxy Server 192 168 66 202 Update gt Outbou
3. Lan To Mobile DES Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot Q C C O1 0C l2 OQ Delete Selected Delete All Add New Position 0 49 CID Ex 0911111111 09117 URL Ex 192 168 0 1 28t When the GSM number of the MV 370 is called this device transfers the call to URL according to the caller ID of the incoming 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 starting with 0911 will be accepted to transfer 3 this means any incoming call is accepted to transfer 4 N this means the incoming call without showing its CID is accepted to transfer d Please note the priority of the routing rules the CID with more digits gets the priority 8 1 2 URL The IP address of destination 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 3 The phone numbers Note If the device has registered at proxy server Asterisk you can enter any destination phone number Also note that in the proxy server Asterisk you need to set the route of destination phone number 4 Leave it blank or N this mean to refuse to transfer 5 this means to transfer via 2 stage dialing The call will be answered with a pr
4. tag as5dee3942 To lt sip 1002 192 168 66 203 5060 gt Contact lt sip Unknown 192 168 66 202 gt Call ID 5ebc2211278e2cb769991 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 z9hG4bK672fa6 7f59c2223275f5ee286d27597a 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 tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502f84042 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 56
5. 1 The complete number e g 0911111111 2 E this means to transfer via 2 stage dialing The call will be answered with a prompt dial tone for the caller to press the destination phone numbers e g 0911111111 3 this allow the caller with lan phone dial directly the destination numbers Precondition 1 MV 370 and incoming lan Phone are both registered at proxy server or Asterisk 2 Proxy server asterisk has set the routing rules to assign specific prefix of numbers to be transferred from MV 370 3 Lan to Mobile routing sets Usage 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 n a ppp this means to do the above routing and to modify the numbers Note d n means to delete the number of prefix a ppp means to add ppp prefix E g d2a09 means to call the registered numbers via one stage dialing The numbers are modified to delete 2 digits of prefix of the original numbers then add 09 to be new prefix of the destination numbers 11 9 Mobile 9 1 Mobile Mobile Status In this page Mobile Status you could get the information of your GSM network and the latest operation PORTech Ed CTI Abt Mobile Status 2008 05 16 14 33 Route Network Registration Mobile SIM Card ID TER Settings Signal Qua
6. DA IM reserare 33 LL 14015161 0 mm 35 17 SETTING AND CHECKING VIA EVR a eio cat o ino bad oe eaae ER ab Sato eo a eu ba pna due nu pnao oi vada n 36 IS SPECIBICA TION 4iecete ias rideo yo io FU ERE PEDVE Sepe d Ceo eV e EY TUE REEF EE PONTE MEE pee Usa 38 19 APPLICATIONS em 39 20 SIMPLE STEPS scc c 40 21 APPENDIX SETUP MV 370 WITH ASTERISK cscssssssssssssssssssssssscssseesesssesoees 41 22 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM 47 1 Introduction MV 370 series products provide you the best connect solution for heterogeneous network including WLAN gt GSM or PSTN You may use a SIP protocol VoIP phone or software to connect to the MV 370 then reach this call to the mobile network and vice versa With multiple sets of MV 370 you may even build an international call 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 operate 2 4 50 sets of LAN gt MOBILE routing and 50 sets of MOBILE gt LAN routing 2 5 Series connections to save bills 2 6 Standard SIP RFC2543 RFC3261 protocol to communicate with other gateways or PC 2 7 settings and managing via web page 3 The contents in package 3 1 MV 370 main body 3 2 AC DC Adaptor 110V AC 12V DC or 220V AC 12V DC 3 3 Network c
7. WAN Setting IP Type Fixed IP o DHCP Client PPPoE IP MV37OIP Mask 255 255 255 0 Gateway Router IP DNS Serverl 168 95 192 1 DNS Server2 168 95 1 1 MAC PPPoE Setting User Name Password Subrnit Reset LAN To Mobile Table Page MEM e NN 0 Your Asterisk IP 1 2 3 4 5 5 7 8 S9 Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM 42 Mobile To LAN Table Page 1 v ltem C D URL Select Authorised Mobile 103 1 Another Authorised Mobile 103 2 3 4 5 6 7 8 The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk These mobile number must be defined as your GSM provider displays the number If you don t know how it is displayed just give a call to the box and check the number given in the Incoming Mob field of the Mobile Status page 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 43 Service Domain Settings Realm 1 Default Active Display Name User Name Register Name Register Password Domain Server Proxy Server Outbound Proxy Status ON O OFF 103 m 103 103 Asterisk Extension Password Asterisk IP Mot Registered Once Asterisk configuration is made you should get Registered on
8. address using numbers on the telephone keypad Use the star key when entering a decimal point 37 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 VAD CNG AEC LEC Packet loss 18 5 GSM MV 370 Dual BAND 900 1800 MHZ Tri BAND 900 1800 1900 MHZ Quad BAND 900 1800 1900 850 MHZ 38 19 Applications 1 Connect to VoipBuster a Register VoipBuster account at Service Domain b Route setting Mobile to Lan set When you call in GSM number of MV 370 you can enter destination number that will dial out from VoipBuster Landline is free GSM rate is cheap 2 How to apply 2 sets of MV 370 1 When you call the no 1 MV 370 gsm number it will provide dial tone and you enter a destination number Then no 2 MV 370 will dial this number and connect Step 1 no 1 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 the prefix of destination number to dial out from no 2 MV 370 2 When you call the no 1 MV 370 gsm number no 2 MV 370 will dial this specific number and connect
9. could change your login name and password PORTech System Authority Your CTI Partner You could change the login usemame password in this page Route New username Mobile New password Network Confirmed password SIP Settings NAT Transform Update System Authority GE Reboot 31 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 PO RTech Save Changes Your CTI Partner You have to save changes to effect them Route Mobile Save Changes Network SIP Settings NAT Transform Update System Authority Save Change Reboot 32 15 Update Here you could update the latest firmware and restore the default settings 15 1 Update New Firmware 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 note no need to unzip the firmware file 4 Click Update it takes few minutes to generate new firmware 5 Please click update default setting after update firmware PO RTech Update Firmware Your CTI Partner You could update the newest firmware PCB mark 2K123C Route Mobile r Method HTTP O TFTP Network SIP Settings NAT Transform Code T
10. dial tone for the caller to press the Num and then the device connects the URL as destination Example after you call the GSM number of the device and hear a dial tone you press 0 then the lan phone of IP address 192 168 0 107 rings 9 8 3 Route LAN to Mobile Settings In this page Lan To Mobile table you could set the routing rules to transfer the calls incoming from Lan to Mobile Maximum 50 sets PORTech aN To Mobile Table Your CTI Partner Route Page c ald aie A item URL Call Num Select Mobile To Lan Speed Dial a o Len To Mabie Senngs Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot oon ont WN O Delete Selected Delete All When the Lan of the MV 370 is called this device transfers the call to Call Num according to the URL of the incoming call 8 3 1 URL The IP address or proxy extension numbers of 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 means the IP address starting with 192 168 0 would be accepted to transfer 4 Part of the proxy extension numbers plus e g 10 This means the extension numbers starting with 10 would be accepted to transfer 10 8 3 2 Call Num the phone numbers of destination You could set the Call Num as the following formats
11. 6f3 192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x Content Length 254 v 0 0 1001 4804366 4807851 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 50 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 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 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 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 0d3b2879 response 8aaaaad5b5ad53 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 0 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite 5 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 teleph
12. ES Gr 32 W RPort Setting Codec Priority 8 G726 40 SIP Responses Other Settings RTP Packet Length NAT Transform G711 amp G729 20ms v Update a G 723 30 ms v System Authority CONTE G 723 5 3K G 723 5 3K OO Off Reboot Voice VAD Voice VAD OO Off 24 11 4 Codec ID Setting You can setup the Codec ID in this page PORTech Codec ID Setting Your CTI Partner You could set the value of Codec ID in this page Route Mobile Codec Type DY Default Value VENUES G726 16 ID 23 g5 255 23 G726 24 ID 22 95 255 22 G726 32 ID 2 95 255 2 SIP Settings Service Domain Port Settings 6726 40 ID 21 95 255 21 RFC 2833 ID MO 95 255 101 Codec ID Setting i DTMF Setting RPart Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 25 11 5 DTMF Setting You can setup the DTMF Setting in this page PORTech DTMF Setting Route Mobile 2933 Network O Inband DTMF 7 Send DTMF SIP Infi SIP Settings O sar id Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot Mobile DTMF debounce 80 range 40 200 default 80 step 10ms Note If this device has registered at SIP Proxy Server Asterisk please select 2833
13. MV 370 VoIP GSM Gateway User Manual PORTech Communications Inc Content Dc IINTRODUC TION sicssesecssscnssocacouacsubaonsvessensseceosnsesessnnaasencsiseosdeonssosssdieuesdunenasensstsasessonsseessuees 1 2 FUNCTIONS eec 1 3 THE CONTENTS IN PACKAGE sicsccssssssscasssesessasessncesscstcausactesdassecsesiossseicssscenddesssecsecansdses 2 4 DIMENSION AND PANEL DESCRIPTION ssesessesseseeseeseeseecesoesereossescesorseeceseeeeseesersessee 3 5 ACCESSORY ATTACHMENT 5uecesdoren keksstivabe rbd eko hoe eka a pris Cue he eue enia epe vd ipae va qU ph 4 6 SETTING AND MANAGING VIA WEB PAGE eere eene enne enne tn ntn asta ano 5 TESYS TEM INFORMATION sss scspscsievivvcccsisevsaccesshcasvesensessenstcacouspuseceeievcssuusestdasteipassseuiuccsueses 6 23 40 BH lj uses TNT 6 PS MIO BULLE omi PU 12 10 NETWORK is cecsssaccstsscccesstiedacsedbatvetuivecesiasodecsteitatoniatisessbessaivsantsdeveaansedeesslbvedschetsssoubuciadens 18 1l SIP SETTING wccssssicstecasscseissseseacatscousansadelssassencsausacsasdacacsasaases bia kai idi bee eii br ee EISE RR 21 12 INA TRAINS cioe avesers sese A Duro oe Deis a upE Cor e FERE FEMA VE FERE E eae FEES TV RR PE CNUEU UR 30 ISSYSTENCAULLH oiaeextessesteeied ben e ke EHE koe EEaee IPLE NN eR Rena s FEVER VERDE MM HER PEE SERLUN RI Hee a URN MEER hives 31 I4 S X VE CHANGD iss sicssscassessessnsssua cabaret sseusdstscassusscasnsdtencisussaucedssesd ciasadiescassses ussebeseacdsscdbsdessen 32 ES UP
14. Step 1 no 1 MV 370 mobile to lan set route table specific destination number 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 the prefix of destination number to dial out from no 2 MV 370 230 20 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 gt it is two stage dialing when mobile call in MV 370 will provide dial tone and you can enter ip or asterisk extension or phone number f 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 will connect with this specific extension or IP or phone number auto f you want to set specific phone number please note your Asterisk need to have route of destination number Lan to Mobile 1 gt it is two stage 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 auto 3 gt It is 1 stage dialing When lan phone and MV 370 both register Asterisk you can dial any destin
15. able 3 4 Antenna 3 5 User s Manual 3 2 Dea 3 4 When you receive MV 370 package and find it is damaged or incorrect please contact your vendor 4 Dimension and Panel description 14 5cm 4 1 4 2 4 5 4 7 4 1 Antenna Antenna connector 4 2 DC 12V Power socket 4 3 LAN Standard RJ 45 socket connecting to Hub circuit 4 4 PWR Power indicator light red light Light is on when system s power supply is normal 4 5 MOBILE GSM indicator light green light Light flashes when GSM status is normal light turns on constantly when GSM is called 4 6 LAN LAN indicator light green light Light flashes when 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 3 5 Accessory attachment 5 1 Connect the network cable both to your Hub and to LAN socket of MV 370 5 2 Connect the antenna and place it in a good receiving location not too close to the device 5 3 Insert a SIM card into back of MV 370 5 4 Plug the adapter in DC 12V socket and PWR socket The PWR light should turn red at the moment 5 5 Click reset button 3 sec MV 370 will restore default IP Other setting as usual LEES fons MAC No 00037E000BCA Cim LIS EI LL reset button 6 Setting and managing via web page The default IP address of MV 370 is http 192 168 0 100 Before accessing the web page p
16. are busying We recommend you to turn this on if you use SIP Proxy 28 11 8 Other Settings You could setup the RFC and QoS according to your ISP information 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 Internet But the QoS function still need to cooperate with the others Internet devices PORTech ol ied Other Settings Route Hold by RFC of Mobile Hold by RFC of Mobile n Mobile Network Voice QoS SIP Settings SIP QoS Service Domain SIP Expire Time Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses NAT Transform Update System Authority Save Change Reboot 29 OoO0n Off OO0n O Off 40 0 63 40 0 63 30 60 86400 sec 12 NAT Trans In this page NAT Trans STUN 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 remember to click the Submit button PORTech sTUN Setting Your CTI Partner BUS STUN of Mobile OOO Off Mobile STUN Server stun xten com Network STUN Port 3478 1024 65535 SIP Settings NAT Transform STUN Setting System Authority Save Change Reboot 30 13 System Auth In this page System Authority you
17. ation number from lan phone directly Please note Asterisk need to set route of destination number that dial out from MV 370 All changes both need to click save and change 40 21 Appendix Setup MV 370 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 lt gsm network gt MV 370 lan Asterisk lt internet gt VOIP provider lt whatever gt landline To do such a call you just call your MV 370 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 for free You can then call all around the world from your 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 into account To have the MV 370 to work with Asterisk you need first to 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 is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation A WAN Settings You could configure the WWAN settings in this page
18. ca93a1e response 4d39ccb0dae64 bb2f1341e9896ac1ea7 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 z9hG4bK590e92b551233a10a0ae71944c19b5aa rec zd 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 192 168 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 SIP 2 0 401 Unauthorized Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae71944c1 9b5aa rec 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 tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502184042 2 192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY WWW Authenticate Digest algorithm MD5 realm asterisk nonce 5def9231 Content Length 0 Scheduling destruction of call 7 45b773130f1fc945efcee502f84042 192 168 66 203 in 15000 ms asterisk1 CLI gt lt SIP read fr
19. e 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 45 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 insecure very qualify yes disallow all allow ulaw prefered codec for DTMF detection allow alaw 21 6 extensions conf eiii GSM Gateway incoming calls gateway exten gt 103 1 Answer 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 sim card mail box thru GSM exten gt _888 1 SetCallerID xxxxxxxxxx exten gt _888 2 Dial SIP EXTEN 103 60 r exten gt 888 3 Hangup 46 22 How to setup Asterisk to receive Caller ID from Test version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a e X Lite 1105x Modify file Add the following setting to etc asterisk sip conf
20. eral Initialization Advanced DTMF Use short headers Expose software version Use obsolete transfer mechanism BY E Also r Restrict caller identity support varies for proxies from different vendors r Use standard status messages otherwise messages will be taken from SIP packets Voice mail number or address V Remove fancy characters from phone numbers rs ee URL Port Fwd to External another device s IP The Explanation of Picture When your MV 370 are busying in line you can set up Fwd to External it s for you to transfer the phone call to another designate device forward to next MV series device 16 9 4 Mobile SMS Agent PO RTech SMS Agent Your CTI Partner Read received SMS Port Status Route Mobile Standby Mobile Status SMS Sender Settings DestNum SMS Agent Maximum Number of UCS2 chars for this text box is 70 Network Message SIP Settings NAT Transform You have 70 UCS2 chars remaining for your description Update System Authority Save Change Reboot 1 Rx List Read received SMS 2 Dest Num the Receiver s phone number 3 Message Here is for you to key the message When you click Rx List you can view all received SMS as follows SMS Rx List Read Status RemotelD 1 REC READ 836936114545 08 01 01 19 34 22 2 REC READ 885935385852 08 03 12 16 25 27 17 Click
21. etwork environment You may refer to your current 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 your Service Provider please input the Username and the Password correctly 5 After you input or modify the value click the Submit button 19 10 3 Network SNTP Settings You could select On to give SNTP 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 synchronize After setting remember to click the Submit button PORTech swrP Settings Your CTI Partner You could set the SNTP servers in this page Route Mobile SNTP Network Primary Server Status WAN Settings Secondary Server SIP Settings Time Zone NAT Transform Sync Time Update System Authority Save Change Reboot 20 On Oof time windows com 208 184 49 9 GMT vi 08 v 00 hh mm ho ho kh d hhimm 11 SIP Setting If you need you could setup the Service Domain Port Settings Codec Settings RTP setting RPort Setting and Other Settings If ISP provides the VoIP service you need to input the related information correctly to register at SIP Proxy Server 11 1 SIP Setting Service D
22. lease confirm this address is available in your network Login VoIP Enter your username and password to login VoIP server Username Password Clear Enter the default username and password to login Default username voip Default password 1234 7 System Information 7 1 After login you could see the system information such as model name firmware version codec version name etc in this page PORTech Mobile VoIP 6 5914 Your CTI Partner Route Model Name MV 370 Mobile Model Description GSM 900 1800 1900MHz T Firmware Version Fri May 16 11 37 26 2008 aishah Codec Version Mon Jul 24 10 55 05 2006 SIP Settings NAT Transform Update System Authority 2007 PORTech Communications Inc Save Change Reboot 7 2 You could also see the setting table in the left side 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 remember to save change whenever you submit any setting Click Save Change then Save button the system will restart and make the changed function setting operative 8 Route 8 1 Route Mobile to LAN Settings In this page Mobile To Lan Table you could set the routing rules to transfer the calls incoming from MOBILE to LAN Maximum 50 sets PORTech fobile To LAN Table Rh CTI Adeste Route Page 1 v Dn
23. lity L Fwd Settings GSM S N IME ttn SMS Agent Network Incoming IP SIP Settings Incoming IP Name NAT Transform Outgoing IP u Update Incoming Mob System Autharity Outgoing Mob Save Change Reboot 1 Network Registration The telecom carrier which the SIM card been registered 2 SIM Card ID SIM card ID 3 Signal Quality Signal quality 4 GSM S N IMEI Number Incoming IP The IP address of the last incoming call from LAN 7 Outgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call from MOBILE 2 3 4 5 6 Incoming IP Name proxy server name 7 8 9 Outgoing Mob The called number of the last outgoing call to MOBILE 12 9 2 Mobile Mobile Setting In this page Mobile setting you could adjust the parameter and click on the option to fit your need You could leave those default value before you had tried the complete operation of this device PORTech Your CTI Partner Route Mobile wd Setting SMS Agent Network SIP Settings NAT Transform Update System Authority Save Change Reboot Mobile Setting D voir tx Gain B 1 2 2 VOIP Rx Gains n _ Qnty 3 LAN Dialtone Gain 9 9 12 4 Routing Range D Ita 49 0 49 7 CODEC Tx Gam 6 7 6 CODEC Rx Gain B 07 8 SIP From Tel User Standard Answer Delay o 0 15 12 9 CLID Presentation Suppression Invocatio
24. n 10 Mobile PIN Code On O Code Confirmed 1 1 LAN Answer Mode Answered Alerted Income 1 VoIP Tx Gain To adjust the volume of LAN side 2 VoIP Rx Gain To adjust the volume of Mobile side 3 LAN Dialtone Gain DTMF Reciver is not good you can adjust gain down 4 ON Off If you use this channel please click on Otherwise please click off 13 5 CODEC Tx Gain as above 6 CODEC Rx Gain as above 7 SIP From Caller ID transfer e Tel 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 22 How to setup Asterisk to receive Caller ID from MV 370 page 43 MV 370 2 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 will send the message as follows in the Packet From caller number sip caller number 192 168 0 228 gt tag 6ac93f7c Please note lf you choose this option please don t register to Asterisk and proxy server Please only fill proxy 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 gt tag 7f130947 lt If you choose this option please don t register to Asterisk and proxy server Please only fill proxy
25. nd Proxy 192 168 66 202 Reboot Status Registered Active COn Of Display Nam User Name Register Nami Register Password Be TITT eee 100 7 pstn gt call 0928492911 mobile number gt MV 370 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 Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt Contact lt sip 1002 192 168 66 202 gt Call ID 20fa417265e6a26d0b0aae4f551 f06f3 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 49 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 z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt tag 677373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f551 f0
26. om 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 54 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK67 2fa6 7f59c2223275f5ee286d27597a From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 3 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 11 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 5def9231 response 046a41 2f4e7ed4 e98fd507416994a80a uri sip 192 168 66 202 algorithm MD5 User Agent CMI CM5K Content Length 0 11 headers 0 lines Using latest REGISTER request as basis request Sending to 192 168 66 203 5060 NAT Transmitting NAT to 192 168 66 203 5060 SIP 2 0 100 Trying Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK672fa6 7f59c2223275f5ee286d27597a 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 3 192 168 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 55 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 lt sip Unknown 192 168 66 202 gt
27. omain 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 PORTech four Cl Posirer service Domain Settings Route Realm 1 Default Active G ON O OFF Mobile Display Name David NONO User Name 5007 Nu SIP Settings Register Name S007 Service Domain Register Password ecco Port Settings r Codec Settings Domain Server ee Codec ID Setting Proxy Server 192 168 0 288 DTMF Setting ee eae RPart Setting italia SIP Responses Status Not Registered Other Settings 1 Active click On to enable the function in Service Domain then input the following items Display name input the name you would like to display User name input your user name in ISP Register Name input your register name in ISP Register Password input your password in ISP Domain Server input the Domain Server IP address 2 3 4 5 6 7 Proxy Server input the Proxy Server IP address 21 8 Outbound Proxy input 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 Status in the Status item Example Register VoipBuster Realm 1 Default Active On Off Display Name jennyQ922 U
28. ompt dial 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 to Lan setting 1 Mobile to Lan 0932 0911123456 When the GSM numbers of the device is called if the caller s prefix numbers are 0932 MV 370 transfers the call to 0911123456 then 0911123456 rings while available Precondition a MV 370 has registered at proxy server Asterisk b The proxy server Asterisk has the route of 09 2 Mobile to Lan Any incoming call gets a prompt dial tone so the caller can enter any IP address sip extension or phone number Precondition a SIP extension or phone number needs to register at SIP Proxy Server or Asterisk b Phone number SIP Proxy Server or Asterisk needs to set the route of destination phone number 8 2 Route Mobile to LAN Speed Dial Settings When you set both Mobile to LAN Speed Dial Settings and Mobile to LAN settings at the same time Mobile to LAN Speed Dial Settings gets higher priority Mobile to Lan setting will be not available PORTech Mobile To LAN Speed Dial LES CTI Asta Route EM LL LL L LL Test 182 158 0 107 Mobile To Lan an Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Delete Selected Delete All Reboot EJ CO SS Gi EX d XS HN E oO The call is answered with a prompt
29. one event 8000 a fmtp 101 0 15 a sendrecv SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 145 7331 branch2 z9hG4bK4C4315351FC84CA582D14FB8C25FC3BF received 192 168 66 145 rport 7331 From user 1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt tag as2a2fbf98 Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt 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 13798 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 52 a fmtp 101 0 16 a silenceSupp off register issue The packet date from Asterisk as follows Please note user_1002 s display name don t appear So the website s Display Name is not available SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK590e92b551233a10a0ae71944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 2 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 10 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 3
30. ser Name jennyQ922 Your Voipbuster username Register Name eny0922 2 2w Register Password 7 Your Voipbuster password Domain Server NENNEN Proxy Server 194 221 62 207 Proxy Server s IP Outbound Proxy Status Reqistered 22 11 2 Port Setting You 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 the Submit button PORTech Ports Setting Your CTI Partner Eus Port of Mobile Mobile SIP Part 5060 1024 55535 Network RTP Port 60000 1024 65535 SIP Settings Sess l Service Domain Port Settings odec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 23 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 PORTech Your CTI Partner Codec Settings Route Mobile Codec Priority 1 G 711 u law v Network Codec Priority 2 G 711 alaw SIP Settings Codec Priority 3 G 723 v Codec Priority 4 G 729 Service Damain aa Port Settings Codec Priority 5 G 726 16 Codec Priority B G 726 24 v Codec ID Setting EN _ DTMF Setting BOSS
31. server ip Username and choose Active on else field empty in sip setting service demain 8 Presentation CLIR If you need to block the Caller Id for call termination please choose Suppression 9 Mobile PIN Code lf you need to unlock pin code via MV 370 you can click On and enter pin code 14 10 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 11 Band Type When you buy Quad band you need to choose your GSM frequency 12 Answer Delay Delay for incoming call when the ring 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 PORTech Forward Setting Your CTI Partner Route C Forward Enable Mobile a FECE COHEN URL Port Status Fwd to External Settings SMS ATET Network SIP Settings NAT Transform Update System Authority Save 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 5 Phone can designate a port which are free to use Profile Options Gen
32. the Realm1 Codec Settings Codec Priority Codec Priority 1 Codec Priority 2 Codec Priority 3 Codec Priority 4 Codec Priority 5 Codec Priority 5 Codec Priority 7 Codec Priority 8 G 711 u law G 711 a law Mot Used Not Used be Mot Used Not Used j Not Usea E Not Used Y RTP Packet Length Le xn ers ene Cif 23 20 ms 30 ms G 723 5 3K taris O On of Voice VAD voice VAD on of 44 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 Mobile Setting VoIP Tx Gain HO 1 12 VoIP Rx Gain 3 15 LAN Dialtone Gain 10 0 12 Mobile ON OFF Routing Range lto 49 0 49 CODEC Tx Gain 6 0 7 CODEC Rx Gain B 7 SIP From Tel User Standard Answer Delay o 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 th
33. the serial no you can view message as follows SMS Reader I C CONNECT RN 886935386862 08 03 12 16 25 27 MY Serial can send SMS and receive SMS Back Delete 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 Network in this page PO RTech Network Status Ede CTI Petree coe cc WAN Interface LAN Interface Mobile Type Fixed IP Client IP 192 168 0 120 Network Mask 255 255 255 0 WAN Settings MAC 00037 E001129 SNTP Settings SIP Settings NAT Transform Update 18 10 2 Network Network Settings WAN 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 your environment PORTech WAN Settings Your CTI Partner You could configure the WAN settings in this page Route z WAN Setting _ Mobile IP Type Fixed IP ODHCP Client O PPPoE Network IP 192 168 0 120 Status Mask 255 255 255 0 WAN Settings Gateway 192 168 0 254 NTP Settings e DNS Server 168 95 192 1 SIP Settings DNS Server2 168 95 1 1 NAT Transform MAC 00037 e001 129 Update System Authority PPPoE Setting Save Change User Name i Reboot Password 1 LAN Mode select NAT 2 Fixed IP the TCP IP Configuration item is to setup the WAN port s n
34. unction Code Action 1 Reboot 195 Reboot the device 2 Factory Reset 198 Return to default settings 3 Check IP Address 120 IVR announces the current IP address Default 192 168 0 100 4 Check IP Type 121 IVR announces DHCP is on or off Default off 5 Check Network 123 IVR announces the current network Mask mask Default 255 255 255 0 6 Check Gateway 2124 2 IVR announces the current gateway IP Address IP address Default 192 168 0 254 7 Check Primary 125 IVR announces the current setting in DNS Server the Primary DNS field Default 192 168 0 1 8 Check Firmware 128 IVR announces the version of the Version firmware 9 Set as DHCP 111 The system is changed to DHCP client Client type 10 Set Static IP 112xxx xx DHCP is disable and system is Address x xxx oodt Changed to static IP type Enter IP address using numbers on the telephone keypad Use the star key when entering a decimal point 11 Set Network Mask 113xxx xx Must set Static IP first 36 X xxx xxxdi Enter value using numbers on the telephone keypad Use the star key when entering a decimal point 12 Set Gateway IP Address 114xxx xx X xxx xxxdi 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 DNS Server 115xxx xx X xxx xxxdi Must set Static IP first Enter IP
35. ype Risc V g Update File Location E ik Default Settings i TFTP Server 192 168 1 250 System Autharity Save Change Reboot Er 15 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 PORTech Restore Default Settings Your CTI Partner You could click the restore button to restore the factory settings Route Mobile Restore default settings Network SIP Settings NAT Transform Update New Firmware Default Settings System Authority Save Change Reboot 34 16 Reboot In this page you could click the Reboot button to restart the system 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 New Firmware Default Settings System Authority Save Change 35 17 Setting and checking via IVR User could get or set some parameters of the system by dialing in the mobile numbers of the device The status or result is reported via voice response system In the first 20 seconds after power on when only Mobile light flash you could dial its mobile numbers When you hear the dial tone press the following codes to set or check the device Item F
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