MV-372 VoIP GSM Gateway User Manual PORTech
Contents
1. Status WAN Settings LAN Settings SNTP Settings SIP Settings NAT Transform Update System Authority Save Change Reboot LAN Settings LAN Setting IP 192 168 0 102 Mask 255 255 255 0 MAC 00037 e008888 DHCP Server OOn Ooff Start IP 150 End IP 200 Lease Time 1 D dd hh 2 11 4 SNTP Settings SNTP Setting function you can setup the primary and second SNTP Server IP Address to get the date time information Also you can base on your location to set the Time Zone and how long need to synchronize again When you finished the setting please click the Submit button PORTech SNTP Settings Your CTI Partner You could set the SNTP servers in this page Route Mobile SNTP On Oof Network Primary Server time windows com Status WAN Settings Secondary Server 208 184 49 9 LAN Settings SNTF Settings Time Zone GMT 08 w 00 v hh mm SIP Settings Sync Time 4 0 0 dd hh mm NAT Transform Update System Authority Save Change Reboot 22 12 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 12 1 In Servcie Domain Function you need to input the accou2. Route Mobile Network SIP Settings Service Domain Port Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update system Authority Save Change Reboot Codec Settings Codec Priority Codec Priority 1 G 711 u law v Codec Priority 2 G711 a law v Codec Priority 3 673 v Codec Priority 4 G72 Codec Priority 5 G 726 16 w Codec Priority 6 G7268 24 w Codec Priority 7 6726 32 M Codec Priority 8 G 726 40 RTP Packet Length G 711 amp G 729 20 ms v G 723 30 ms M G 723 5 3K G 723 5 3K Oon Off Voice VAD Voice VAD OoOn Off 26 12 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 D Default Value Network 6726 16 ID 23 95 255 23 G726 24 ID 22 95 255 22 G726 32 ID 2 m 2 Service Domain li Y Port Settings 6726 40 ID 21 95 255 21 Codec Settings REC 2633 ID LN 95 255 101 DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot SIP Settings D 12 5 DTMF Setting You can setup the DTMF Setting in this page PORTech E CTI i Route Mobile Network SIP Settings Serv
3. PORTech Mobile To LAN Speed Dial Your CTI Partner Route mE Test 182 158 0 107 oO Mobile To Lan Settings Mobile To Lan Speed Dial 3 o Mubile Settirm Mobile Network SIP Settings NAT Transform Update System Authority Save Change Delete Selected Delete All Reboot E Oo mn c NES a iu BO es C The call will be answered and prompt dial tone again When the caller may enter the Num 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 9 9 3 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 ES CTI add Route Page it Hobie Tolan Seinas M A ET ed Dia pr To Mobile em Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot EO OOo LIS 0C a d Fe B3 eS C3 The MV 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 10 Call Num 1 may enter the whole number e g 0911111111 2 a simple means 2 stages dialing The call will be answered and prompt dial tone again to receive the called number as th
4. WWwW 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 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 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 7e45b773130f1fc945efcee502f84042 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 54 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 7e45b773130f1fc945efcee502f84042 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 line
5. 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f84042 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 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 z9hG4bK590e92b551233a10a0ae7 1944c19b5aa 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 Call ID 7e45b773130f1fc945efcee502184042 3 192 168 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 zd Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae7 1944c19b5aa 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
6. Network Mask 123 IVR will announce the current network mask Default 255 255 255 0 Check Gateway IP Address 124 IVR will announce the current gateway IP address Default 192 168 0 254 Check Primary 125 IVR will announce the current 38 DNS Server setting in the Primary DNS field Default 192 168 0 1 Check Firmware Version 1 28 IVR will announce the version of the firmware running Set as DHCP client 111 The system will change to DHCP Client type 10 Set Static IP Address 1 12XXX XXX XXX xxx DHCP will be disabled and system will change to the Static IP type Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point 11 Set Network Mask 113XXX XXX XXX xxx Must set Static IP first Enter value using numbers on the telephone key pad Use the star key when entering a decimal point 12 Set Gateway IP Address 114xxx XXX XXX xxx Must set Static IP first Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point 13 Set Primary DNS Server 115xxx XXX XXX oodt Must set Static IP first Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point 39 19 Specification 19 1 Pro
7. Transform Incoming IP Name Update Outgoing IP System Authority Save Change Incoming Mab Reboot Outgoing Mob 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 5 Incoming IP The IP address of the last incoming call from LAN 6 Incoming IP Name proxy server name 7 Outgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call from MOBILE 9 Outgoing Mob The called number of the last outgoing call to MOBILE 12 10 2 Mobile Setting PORTech LA CTI esl Route Mobile Status Settings Fwd Settings SMS Agent Network SIP Settings NAT Transform Update System Authority Save Change Reboot ae 1 VoIP Tx Gain Ay 2 VoIP Rx Gain Mobile Setting 1 vaip Tx Gain 9 2 VoIP Rx Gain 11 9 12 0 15 3 LAN Dialtone Gain 3 p 12 4 Mobile 1 ON OFF 5 Routing Range Do J to 43 0 49 6 CODEC Tx Gain e p 7 7 CODECRxGain 6 a7 8 SIP From Tel User Standard Answer Delay 0 0 15 12 9 CLID Presentation Suppression Invocation 10 Mobile PIN Code On Code Confirmed 11 LAN Answer Mode Answered Alerted Income Routing Range lto 49 0 49 CODEC Tx Gain e p 7 CODEC Rx Gain 6 7 SIP From Te
8. function you want to set up PO RTech Mobile VoIP2 6 514 Route Model Name MV 372 Mobile Model Description GSM 900 1800MHz HO Firmware Version Fri May 16 11 30 35 2008 EN Codec Version Mon Jul 24 10 55 05 2006 SIP Settings NAT Transform Update System Autharity 2007 PORTech Communications Inc Save Change Reboot 9 Route Important The route table 50 sets can share by two channels The setting please refer 10 2 Mobile setting ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for item 25 49 9 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 Page E eM es P SETEC L1 Mobile To Lan Settings viobile To Can Speed Dia Lan To Mobile Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot oon Oa fF C N O Delete Selected Delete All reset Add New Position 0 49 CID Ex 0911111111 0911 URL Ex 192 158 0 1 28t The MV 372 will transfer to the URL according to the caller ID of the Mobile CID 1 may enter the whole number e g 0911111111 2 only part of the number prefix e g 0911 means any number starting with 0911 will be accepted 3 means all numbers can be accepted d 4 N means the calls without the CID Please note the priority of the rules The item which
9. 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 N it means refuse to transfer 3 If an 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 0 101 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 372 have register proxy server Asterisk The proxy server Asterisk have the route 09 When the callers prefix number is 0932 MV 372 will connect 0911123456 automaticlly 2 Mobile to Lan Any caller call the MV 372 s sim 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 Proxy Server or Asterisk Phone number SIP Proxy Server or Asterisk need to set the route of this phone number 9 2 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time MV 372 will give priority to Mobile to LAN Speed Dial Settings
10. 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 PO RTech WAN Settings Your CTI Partner You could configure the WAN settings in this page Route Network Made O Bridge NAT Mobile Network WAN Setting IP Type Fixed IP ODHCP Client O PPPoE atus WAN Settings IP 192 168 0 122 LAN Settings Mask 255 255 255 0 SNTP Settings Gateway 192 168 0 254 PME DNS Server 168 95 192 1 NAT Transform DNS Server2 168 95 1 1 Update MAC 00037 e009999 System Authority i Save Change PPPoE Setting Reboot User Name Password E o 11 3 LAN Settings You can check the current Network setting in this page 20 1 The TCP IP Configuration item is to setup the WAN port s network environment You may refer to your current network environment to configure the system properly 2 DHCP Server You may refer to your current network environment to configure the system properly PORTech Your CTI Partner Route Mobile Network
11. 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 f06f3 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 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 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 Test ok INVITE sip 1002 192 168 66 202 SIP 2 0 50 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 8aaaaa5b5ad53 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 2
12. 54 v 0 07 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101 a rtpmap 0 pcmu 8000 a rtpmap 8 pcma 8000 a rtpmap 3 gsm 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 145 7331 branch z9hG4bK4C4315351F C84CA582D14FB8C25FC3BF sreceived 192 168 66 145 rport 7331 5 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 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 lt 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 52 To lt sip
13. MV 372 VoIP GSM Gateway User Manual PO Caschi rech PORTech Communications Inc Content LIN TRODUC TION rerom 1 2 FUNCTION DESCRIPTION scssserccisdrsiseiesinceseiaesisdeusevesousaenesudtcndeSbssousudsechesesessacitursnaasssedes 1 SA ga 34 Ih JE 8I NA teste 1 Cb y Eos T L piste c 2 S CHART OF THE DEVICE sestssssacsssstiuascapeusstaboiaboatageseesessaseneanssavaesusauwagatennenssstbiesas ANE MER 3 G CABLING E EE E E E E EE EEA O EE EAA 4 TWEB PAGE SETTING ssisessssechenaisa coe savtssdsensalsasubactovaigucnenssbaceiuebselonsanietsdedsausesssbincesbaskansased 5 S SYSTEM INFORMATION scscusessscedbesessdivessacesinteassinsssbasensseosessesnseustebsisushenwanssunessvaeupesesadtons 6 Ds ROUT uses 6 PO MOBI porre 12 LLUNEDWOR so orctscconscsundensateseuseassieubonssdsuscedetsounedbncantesusnsakeshisbodiuesenevedosaseuse asdunesienbenesovatanisen 19 DZ SEP SE DEINGE es iicsasenesecsssceccocteouscenkeonssbecwessoos tungeasepenseateossnsvsbacovasosvennbvestenseatecaeestebesannuxeeeen 23 I3 NAT ERUANNS ibid i E etvao Piae see dope e COPA TREER QR oM Via UPS CHOIR Oa eR DURITIA TRUE 32 IA SVSTEM AUTH iiri eH EET n vy ao SU YER HET HIE EEA E A IE ENTERA c EVE e ME 33 15 SA VE CHANGE i nss eseitedeekebviu eaten elec b o ive pkME eise eb deer bist eive eie sas Rx Va en PEEL Yin Sias sosea isesi 34 FOLD P D HU Ofen 35 PT 30111018 d inesset 37 IS IP SELLING secs E aet ie ER E dead eye io Feb aod heated iib dri
14. a HEU RIEN CI IMPARARE SERRE 38 19 SPECIFICA TION ensem 40 20 APPENDIX SETUP MV 370 WITH ASTERISK eere eee eene ettet nennt 41 21 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM MV 37 47 PA TRI I BLUR I M UI us Feo 57 1 Introduction MV 372 is a 2 channels VolP 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 2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone 2 Function description 2 1 VoIP SIP GSM 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 2 4 Series connections to save bills 2 5 Standard SIP RFC2543 RFC3261 protocol 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 372 main body 3 2 Power adaptor AC DC 110V AC 12V DC or 220V AC 12V DC 3 3 Network cable 3 4 Antenna 3 5 User Manual 2 4 4 Dimension 5 Chart of the device 5 1 ntenna 5 2 5 3 5 4 5 5 5 6 5 75 8 5 1 Antenna Antenna connector 5 2 DC 12V Power input 5 3 LAN LAN port It also can be DHCP Server 5 4 WAN RJ 45 internet connector gt standard RJ 45 socket connect to HUB 5 5 PWR Power LED Light up when power i
15. e destination e g 0911111111 or 0911111111 3 d n a ppp for one stage dialing is option d n means to delete the beginning n codes a ppp means to add ppp 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 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 372 will connect this call auto Example of Application When you call the ch 1 MV 372 gsm number it will provide dial tone and you enter a destination number Then ch 2 MV 372 will dial this number and connect ch 1 MV 372 mobile to lan set route table ch 2 MV 372 lan to mobile set route table Additionally two channels 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 372 The channel 2 MV 372 s ip the first ip 5062 e g http 192 168 0 100 5062 1 10 Mobile 10 1 Mobile Status PO RTech Mobile Status Your CTI Partner 2008 05 15 18 10 Route Mobile Network Registration Chunghwa e SIM Card ID EE Fwd Settings SMS Agent Signal Quality 17 Network GSM S N aia SIP Settings Incoming IP NAT
16. e call in 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 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 it is two stage dialing when lan phone call in MV 372 will provide dial tone and you can enter mobile number 2 specific mobile number when lan phone call in MV 372 will connect with the specific mobile number auto 3 gt It is 1 stage dialing When lan phone and 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 372 All changes both need to click save and change 57 58
17. ese functions can help your VoIP device working properly behind NAT 13 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 I CTI esiste STUN of Mobile 1 Oon Off STUN of Mobile 2 OOn Off Route Mobile Network STUN Server SIP Settings STUN Port 34780 1024 5535 NAT Transform Update System Authority Save Change Reboot 32 14 System Auth In System Authority you can change your login name and password PORTeCh System Authority S CTI Jis You could change the login username password in this page Route New username Mobile New password Network Confirmed password SIP Settings NAT Transform Update System Authority Save Change Reboot 33 15 Save Change In Save Change you can save the changes you have done If you want to use new setting in the VoIP system You have to click the Save button After you click the Save button the system will automatically restart and the new setting will effect PORTech gave Changes Your CTI Partner You have to save changes to effect them Route Mobile cave Changes Network SIP Settings NAT Transform Update New Firmware Default Setti
18. ice Domain Port Settings Codec Settings Codec ID Setting RPart Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot DTMF Setting Mobile DTMF Transfer to Lan 9 2833 Inband DTMF Send DTMF SIP Info Mobile DTMF debounce EJ 28 range 40 200 default 80 step 10ms 12 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 PO RTech RPort Setting Your CTI Partner Route RPort of Mobile 1 9 On O Of Mobile RPart of Mobile 2 90n O Off Network Submit Reset SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update system Authority Save Change Reboot 29 12 7 SIP Responses PO RTech SIP Responses Setting E CTI Praise Route Response on port busy Mobile 9 486 Busy here Network 503 Service unavailable _SIP Settings SIP Responses Service Domain ON QOOFF 180 Ringing Auto force to ON if 183 was OFF Port Settings OON QGOFF 183 Session Progress Codec Settings Codec ID Setting rius Setting Por erung SIP Responses er Settings NAT Transform Update System Authority Save Change Reboot 12 7 1 486 busy here 503 Service unavailable When Device a
19. ile 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 20 4 Asterisk configuration Once the MV 372 is set you have to configure Asterisk On that side you have to setup files as follow 20 5 sip conf GSM VOIP Gateway MV 372 103 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 20 6 extensions conf 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 372 sim card mail box thru GSM exten gt _888 1 SetCallerID xxxxxxxxxx exten gt _888 2 Dial SIP EXTEN 103 60 r exte
20. l User Standard Answer Delay o 0 15 CLID Presentation Suppression Invocation Mobile PIN Code On Code Confirmed LAN Answer Mode Answered Alerted Income Mobile 1 6 Rx Mobile 2 R 1 VoIP Tx Gain To adjust the volume of LAN side 2 VoIP Rx Gain To adjust the volume of Mobile side 13 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 5 Routing Range The 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 CODEC Rx Gain as above 8 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 21 How to setup Asterisk to receive Caller ID from MV 372 page 42 MV 372 will send the message as follows in the Packet From caller number lt sip 3001 192 168 0 228 gt tag 51088abb Tel Tel MV 372 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
21. n gt 888 3 Hangup 46 21 How to setup Asterisk to receive Caller ID from MV 372 Test version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a X Lite 1105x Modify file Add the following setting to etc asterisk sip conf 1000 type friend secret 1000 qualify yes nat yes host dynamic canreinvite no context internal 1001 type friend secret 1001 qualifyzyes nat yes host dynamic canreinvite no context internal 1002 type friend secret 1002 qualify yes 47 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 372 address 192 168 66 203 5060 username 1002 displayname user_1002 oo v e http 192 168 66 203 1ogin cgi gsx Search y BRO SEO HAO SHREW IAM RAW Erehe Sp ase 3 e a amp Ve SEP i VoIP Web Management Mobile Voi Service Domain Settings You could set information of service domains in this page Route gt Mobile No Mobile 1 Network Realm 1 Default Active On Of SIP Settings gt Di
22. ngs system Authority Save Change Reboot 34 16 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 16 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 lich b Update Firmware You could update the newest firmware PCB mark 2K123B Route Mobile Method HTTP TFTP Network Lo NAT Transform Code Type Rise v Update File Location Default Settings Cerea eee i TFTP Server 192 168 1 250 System Authority Save Change U pdate Update Reset 35 16 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 b CTI abd You could click the restore button to restore the factory settings Route Mobile Restore default settings Netw
23. nt 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 203 PORTech Your CTI Partner Route Mobile Network SIP Settings ervice Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Example Service Domain Settings Mobile 1 v Realm 1 Default Active Dis
24. ork SIP Settings NAT Transform Update New Firmware Default Settings System Authority Save Change Reboot 36 17 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 PO RTech Reboot System E CTI minae You could press the reboot button to restart the system Route Mobile Reboot system Network SIP Settings NAT Transform Update System Authority Save Change 37 18 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 Item IVR Action IVR Menu Choice Notes 1 Reboot 1 95 After you hear Option Successful hang up Unit will reboot automatically Factory Reset 198 System will automatically Reboot WARNING ALL User Changeable NONDEFAULT SETTINGS WILL BE LOST This will include network and service provider data Check IP Address 1 20 IVR will announce the current IP address f 192 168 0 100 Default Check IP Type 121 IVR will announce if DHCP in enabled or disabled default OFF Check
25. play Name User Name Register Name Register Password Domain Server Proxy Server Outbound Proxy Status Register VoipBuster Active Display Name User Name Register Name Register Password Domain Server Proxy Server Outbound Proxy Status On Off ON O OFF 3001 3001 3001 B1 218 151 230 Not Registered jennyQ922 jennyQ922 Your Voipbuster username fjenny0922 n Your Voipbuster password 194 221 52 207 Proxy Server s IP Reqistered 24 12 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 PU RTech Ports Setting Your CTI Partner o Mobile SIP Port 5060 1024 65535 Network RTP Port 60000 1024 66535 es Service Domain SIP Port 5062 1024 65535 Duas errr RTP Port 60100 1024 65535 Codec ID Setting DTMF Setting RPort Setting SIP Responses _Other Settings NAT Transform Update system Authority Save Change Reboot 25 12 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 PORTech Your CTI Partner
26. re busying you can select 486 or 505 to response to SIP 12 7 2 180 Ring on off LAN TO MOBILE two stage dialing can be turn off therefore there will be no the Ring Back Tone all the phone call will be transferred to Voice Mail directly For this function 183 must be turn on 12 7 3 183 Session Progress gt It means on progressing When you turn 183 on it means you can hear voicemail while GMS side are busying We recommend you to turn this on if you use SIP Proxy 30 12 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 0 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 PORTech Other Settings Your CTI Partner Route Hold by RFC of Mobile 1 OO Off Hold by RFC of Mobile 2 Oon Off Mobile Network Voice QoS 40 0 63 SIP Settings SIP QoS 40 083 Service Domain SIP Expire Time 30 60 86400 sec Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 31 13 NAT Trans In NAT Trans you can setup STUN and uPnP function Th
27. re are some screen shots showing all the important parameters You have to note that in all the configuration process the 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 WWAN settings in this page WAN Setting IP Type Fixed IP C DHCP Client C PPPoE IP M V37D IP Mask 255 255 255 0 Gateway Router iP DNS Servert 168 95 192 1 DNS Server2 158 95 1 1 MAC PPPoE Setting User Name Password Submit Reset Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM LAN To Mobile Table Page 1 a Your Asterisk IP F1 1 3 4 6 i 8 3 42 Mobile To LAN Table Page 1 he tem ee Select Authorised Mobile 103 1 Another Authorised Mobile 103 2 3 4 5 6 7 8 9 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 Defa
28. rt Status Route Mobile 1 Standby Mobile Mobile 2 Standby Status oe SMS Sender SMS Agent Via Mobile 1 O2 Network Dest Num Maximum Number of UCS2 chars for this text box is 70 SIP Settings NAT Transform Message Update System Authority You have 70 UCS2 chars remaining for your description Save Change Reboot 1 Rx List Read received SMS 2 Dest Num the Receiver s phone number 3 Message Please fill the message that want to send to receiver When you click Rx List you can view all received SMS as follows SMS Rx List Read Status l RemotelD 3 REC READ 886936114545 08 01 01 19 34 22 2 REC READ 896935386862 08 03 12 16 25 27 Click the serial no you can view message as follows 18 SMS Reader D 896935386862 08 03 12 16 25 27 MV Serial can send SMS and receive SMS Back Delete 11 Network In Network you can check the Network status configure the WLAN Settings LAN Setting and SNTP settings 11 1 Network Status You can check the current Network setting in this page Elec oe Network Status Kee Ethernet0 NUT OTT NN Mobile Type Fixed IP Client Fixed IP Client IP 192 168 0 122 192 168 0 102 Network Mask 255 255 255 0 255 255 255 0 Gateway 192168024 1921880254 WAN Seffinne MAC 00037 009999 00037 E008888 11 2 WAN Settings You can check the current Network setting in this page 19 1 The TCP IP Configuration item is
29. s 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 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 55 Transmitting NAT to 192 168 66 203 5060 SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK672fa67f59c2223275f5ee286d27597a 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 22 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 mobil
30. s normal 5 6 VoIP1 an indicator light of VoIP1 5 7 VoIP2 an indicator light of VoIP2 5 8 LINK Indicator Light up when network is connected EN 6 CABLING 6 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 6 2 Connect the antenna and put it in proper position to get the best signal reception 6 3 Insert the SIM card from back of the main body take the slide off first 6 4 Click reset button 3 sec MV 372 will restore default IP Other setting as usual y 6 5 Connect the power adaptor The POWER LED should be light up 7 Web Page Setting When the IP setting is done the operator may setup all the rest parameters via web page Browse the IP address from Internet Explorer e g http 192 168 0 100 The following page shows up Login PORTech VoIP Enter your username and password to login VoIP server Username Password TE o 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 8 System Information 8 1 When you login the web page you can see the demo system current system information like firmware version company etc in this page 8 2 Also you can see the function lists in the left side You can use mouse to click the
31. service demain User Tel MV 372 will send the message as follows in the Packet From Username sip caller number 192 168 0 228 gt tag 7f130947 X If you choose this option please don t register to Asterisk and proxy server Please only fill proxy server ip Username and 14 choose Active on 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 Code If you need to unlock pin code via 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 10 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 5 PO RTech Forward Setting Bb CTI nadie Route Mobile Se cm URL Port Status Fwd to Mobilet 192 168 0 100 5060 Fwd to Mobile2 182 168 0 100 5062 C Forward Enable Fwd Settings SMS Agent Fwd to External Network SIP Settings NAT Transform Update System Au
32. splay Name user 1002 User N 1002 NAT Trans 5 S l Register Name 1002 System Auth Register Password esee Saye Change Domain Server 192 168 66 202 Proxy Server 192 168 66 202 Update gt Outbound Proxy 192 168 66 202 Reboot Status Registered Active On Of Display Name User Name Register Name Register Password Bes LITT TT ees 10 7 48 test1 pstn gt call 0928492911 mobile number gt 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 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 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 49 SIP 2 0 200 Ok Via SIP 2 0 UDP 192 168 66 202 5060 branch
33. thority 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 Phone can designate a port which are free to use 16 Profile Options Use short headers Expose software version Use obsolete transfer mechanism BYE Also 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 MV Remove fancy characters from phone numbers ee Name URL Port Fwd to Mobile1 E c c Fwd to Mobile2 p meos 7 Fwd to External PO 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 you can set up Fwd to External then you can transfer the phone call to another designate device d 10 4 Mobile SMS Agent PORTech SMS Agent Your CTI Partner Read received SMS Po
34. tocols SIP RFC2543 RFC3261 19 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 19 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 19 4 Voice Quality VAD 40 CNG AEC LEC Packet loss 19 5 GSM 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 20 Appendix Setup MV 372 with Asterisk 20 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 372 lt lan gt Asterisk lt internet gt VOIP provider lt whatever gt landline To do such a call you just call your 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 372 for free You can then call all around the world from your mobile at voip cost 20 2 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 have all the parameters taken into account To have the MV 372 to work with Asterisk you need first to configure the A box He
35. ult Active ON O OFF Display Name 103 m User Name 103 Register Name 103 Register Password Domain Server Asterisk IP Proxy Server Outbound Proxy Status Not Registered Once Asterisk configuration is made you should get Registered on the Realm1 Codec Settings Codec Priority Codec Priority 1 G 711 u law v Codec Priority 2 G 711 a law v Codec Priority 3 Mot Used v Codec Priority 4 Not Used v Codec Priority 5 Not Used I Codec Priority 5 Not Used v Codec Priority 7 Not Used Ww Codec Priority 8 Mot Used Ww RTP Packet Length G 711 amp G 729 20 ms r2 2g 30 ms G 723 5 3K if 2o Sess On of Voice VAD Voice VAD O On Off 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 20 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 Mob
Download Pdf Manuals
Related Search