MV-374 / MV-378 VoIP GSM Gateway User Manual PORTech
Contents
1. Route Mobile Network SIP Port 5060 102465535 SIP Settings ile 80000 1024 65535 Service Domain Port of Mobile N E nas SIP Port 5062 1024 65535 Codec Settings M Codec ID Setting RTP Port 60100 1024 65535 DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot de 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 When you finished the setting please click the Submit button PORTech Your CTI Partner Codec Settings Route Codec Priority Mobile Codec Priority 1 G 711 ulaw V Network Codec Priority 2 G711 a law M SIP Settings Codec Priority 3 6 733 z Codec Priority 4 G 729 v Service Domain SA P i Codec Priority 5 G 726 16 M Ct nas Codec Priority B GJ26 24 M Codec ID Setting mem DTMF Setting Codec Priority 7 G 726 32 RPort Setting Codec Priority 8 G 726 40 SIP Responses Other Settings RTP Packet Length NAT Transform G 711 amp G 729 20 ms M Update G 723 30 ms System Authority Save Change G 723 5 3K 6 23 5 3K O On Off Reboot Voice VAD Voice VAD O On Off 28 11 4 Codec ID Setting You can setup the Codec ID in this page PORTech Codec ID Setting Your CTI Partner You2. 7e45b773130f1fc945efcee502f84042 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 z9hG4bK672fa67f59c2223275f5ee286d27597a 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 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 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 Call ID 7e45b773130f1fc945efcee502f84042 192 168 66 203 CSeq 11 REGISTER 57 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 OPTIONS sip 1002 192 168 66 203 5060 SIP 2 0 Via SIP 2 0 UDP 192 168 66 202
3. Mobile 1 2 Route Mobile p A r 1 VoIP Tx Gain 9 0 12 2 VoIP Rx Gain 11 0 15 Status 3 LAN Dialtone Gain 9 Si 3 ialtone Gain O 12 Fwd Settings SMS Agent Mobile 1 ON OFF Network 4 Routing Range 0 to 24 0 49 SIP Settings 5 CODEC Tx Gain e 7 6 CODEC Rx Gain B 7 NAT Transform 7 SIP From Tel User Standard v Answer Delay 0 o 15 8 Update zn 9 CLID Presentation Suppression Invocation System Authority 10 Mobile PIN Code On Code Confirmed Save Change 11 LAN Answer Mode Answered Alerted Income Reboot 12 mobite 2 BON ore Jo Routing Range 25 to 49 0 49 CODEC Tx Gain 6 0 7 CODEC Rx Gain 6 0 7 SIP From Tel User Standard v Answer Delay D 0 15 CLID Presentation Suppression Invocation Mobile PIN Code On Code Confirmed LAN Answer Mode Answered Alerted Income Mobile 1 UTE RET d 1 VoIP Tx Gain At 2 VoIP Rx Gain Mobile 2 Rx 13 1 VoIP Tx Gain To adjust the volume of LAN side 2 VolP Rx Gain To adjust the volume of Mobile side 3 LAN Dialtone Gain DTMF Reciver is not good you can adjust gain down 4 Routing Range The route table 50 sets can share by two channels 1 2 ch 3 4 ch 5 6 ch 7 8 ch ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for item 25 49 B CODEC Tx Gain as above 6 CODEC Rx Gain as above 7 SIP From
4. qualify yes disallow all allow ulaw prefered codec for DTMF detection allow alaw 19 6 extensions conf 48 kkkkkkkkkk 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 Response Timeout 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 374 MV 378 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 49 20 How to setup Asterisk to receive Caller ID from MV 374 MV 378 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 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 50 secret 1002 qualifyzyes 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 19
5. 2 3 or 4 25 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 Example Register VoipBuster Realm 1 Default Active On C Of Display Name ljenny0922 User Name jenny0922 Your Voipbuster username Register Name enny09222 Register Password 7 Your Voipbuster password Domain Server Proxy Server 194 221 62 207 Proxy Server s IP Outbound Proxy NEN Status Registered 26 11 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 PORTech Ports Setting Your CTI Partner You could set the port number in this page
6. Slave 1 192 168 0 112 40000 Status Slave 2 192 168 0 114 40000 Bo Slave 3 1921680 116 40000 SNTP Settings Slave Setting SIP Settings NAT Transform Update System Authority Save Change Reboot 24 11 SIP Setting In SIP Setting you can setup the Service Domain Port Settings Codec Settings RTP setting RPort Setting and Other SettingS If the VoIP service is provided by ISP you need to setup the related informations correctly then you can register to SIP Proxy Server correctly 11 1 In Servcie Domain Function you need to input the account and the related informations in this page please refer to your ISP Provider 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 PORTech MA Service Domain Settings Route Mobile 1 v Mobile Realm 1 Default Network Active ON O OFF SIP Settings Display Narne BE i User Name 803 Register Name 803 Codec Settings Register Password eee Codec ID Setting DTMF Setting Domain Server RPort Setting Proxy Server 192 168 0 1 SIP Responses Other Settings Outbound Proxy NAT Transform Statusi Registered e System Authority Active O ON OFF Save Change Display Name MEM Reboot User Name Register Name First you need to click Active to enable the Service Domain then you can input the following items 1 Choose Mobile 1
7. 5060 branch z9hG4bK 7b92dd8a 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 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 58 21 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 374 MV 378 will provide dial tone and you can enter ip or asterisk extension or phone number If you want to enter phone numb
8. 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 374 MV 378 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 ON Off If you use this channel please click on Otherwise please click off 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 5 PORTech Forward Setting Your CTI Partner Mobile 1 2 v Route Mobile C Forward Enable Status Settings RS ES TN Fwd Settings Fwd to Mobilel EEA Fwd to Mobile2 Network Fwd to External SIP Settings NAT Transform Update System Authority Save Change Reboot ii 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 General Initialization DTMF Use short headers v Expose software version Us obs
9. could set the value of Codec ID in this page Route Mobile 1 Codec Type UID Default Value Network G726 16 ID 23 95 255 23 SIP Settings G726 24 ID 22 _ 85 255 22 Service Domain SRE imena 2 Port Settings G7 26 40 ID 21 95 255 21 a Settin d RFC 2833 ID 101 95 255 101 RPor Seting SIP Responses Other Settings NAT Transform Update _ System Authority Save Change Reboot 29 11 5 DTMF Setting You can setup the DTMF Setting in this page PORTech DTMF Setting Your CTI Partner Route Mobile O 2833 Network Inband DTMF Send DTMF SIP Infi SIP Settings usa ae Service Domain Mobile DTMF debounce 80 range 40 200 default 80 step 10ms Port Settings Codec Settings RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 30 11 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 Rech RPort Setting Mobile 1 2 M Route Mobile TRE RPort of Mobile 1 On O Off RPort of Mobile 2 On O Of SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting R Port Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 31 11 7 SIP Res
10. 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 374 MV 378 for free You can then call all around the world from your mobile at voip cost 19 2 MV 374 MV 378 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 374 MV 378 to work with Asterisk you need first to 43 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 374 MV 378 is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation LAN Settings You could configure the LAN settings in this page LAN Mode C Bridge NAT WAN Setting IP Type IP Mask Gateway DNS Server DNS Server MAC Fixed IP C DHCP Client C PPPoE mw370IP Router IP fees mE 44 LAN To Mobile Table Page z nem O URL Call Num Select 0 your asterisk IP B m 1 H 2 E 3 F 4 E 5 z 5 jaj 7 E B E 9 pz Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM Mobile To LAN Table c Page hs RE NENNEN NU K ZS authorised mobile n 103 LI 1 another authorised n 103 a 2 p 3 F 4 Ez 5 pr 6 F 7
11. 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 means 2 stages dialing The call will be answered and prompt dial tone again to receive the called number as the destination e g 0911111111 or 09111111112 3 Z 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 374 MV 378 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 374 MV 378 will connect this call auto Example of Application When you call the ch 1 MV 374 MV 378 gsm number it will provide dial tone and you enter a destination number Then ch 2 MV 374 MV 378 will dial this number and connect ch 1 MV 374 MV 378 mobile to lan set route table ch 2 MV 374 MV 378 lan to mobile set route table Additionally two channels MV 374 MV 378 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 374 MV 378 10 MV 374 MV 378 s IP
12. 00037 2000003 20 10 2 WAN Settings You can check the current Network setting in this page PORTech wax Settings Your CTI Partner You could configure the WAN settings in this page Route Ethernet 0 v Mobile Network Network Mode Bridge ONAT V CET VAN Settings ka Ee nm IP T LAN Settings ype Fixed IP DHCP Client O PPPoE SNTP Settings IP 192 168 0 110 Slave Setting Mask 255 255 255 0 SIP Settings Gateway 192 168 0 254 NAT Transform DNS Server 168 95 192 1 Update DNS Server2 168 95 1 1 System Authority MAC 00037 e005555 Save Change Reboot User Name Password 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 setup 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 21 10 3 LAN Settings You can check the current Network setting in this page PORTech LAN Settings Eh CTI a _Route Ethernet 0 Mobile KU LAN Setting Setting IP 192 168 0 101
13. 1 gt tag 677373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x 52 Content Length 254 v 0 0 1001 4804366 4807851 IN IP4 192 168 66 145 s X Lite czIN 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 374 MV 378 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 8aaaaa5b5ad53 654bf0a2ab0fa9bb118 uri sip 1002 192 168 66 202 algorithm MD5 Max Forwards 70 Content Type application sdp User Agent X Lite release 1105x Content Length 254 v 0 0 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite czIN IP4 192 168 66 145 t 0 0 m audio 8000 R
14. 2 168 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 displaynamezuser 1001 MV 374 MV 378 address 192 168 66 203 5060 username 1002 displayname user 1002 test1 pstn gt call 0928492911 mobile number gt MV 374 MV 378 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 5 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 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 66 145 733
15. 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 55 11 headers 0 lines Using latest REGISTER request as basis request Sending to 192 168 66 203 5060 NAT Transmitting NAT to 192 168 66 203 5060 SIP 2 0 100 Trying Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae71944c19b5aa rec eived 192 168 66 203 rport 5060 From 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 z9hG4bK590e92b551233a10a0ae71944c19b5aa 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 7e45b773130f1fc945efcee502f84042 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 56 Scheduling destruction of call
16. 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 374 MV 378 page 42 MV 374 MV 378 will send the message as follows in the Packet From caller number lt sip 3001 192 168 0 228 gt tag 51088abb User User Standard If you need to register to Asterisk and proxy server please choose this option MV 374 MV 378 will send the message as follows in the Packet From 3001 lt sip 3001 192 168 0 228 gt tag 51088abb Tel Tel MV 374 MV 378 will send the message as follows in the Packet From caller number sip caller number 192 168 0 228 gt tag 6ac93f7c zt Please note If 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 14 User Tel MV 374 MV 378 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 choose Active on else field empty in sip setting service demain 8 Answer Delay Delay for incoming call when the ring 9
17. E 8 r 9 E The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk 45 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 Service Domain Settings You could set information of serice domains in this page Realm 1 Default Active 6 On C Of Display Name 103 User Name ho Register Name 103 Register Password Asterisk extension password _ Domain Server Proxy Server Asterisk IP Outbound Proxy Status Registered Once Asterisk configuration is made you should get Registered on the Realm1 46 Codec Settings You could set the codec settings in this page Codec Priority Codec Pnority 1 6711 ulew v Codec Priority 2 5711 mow z Codec Pnority 3 NotUsed Cadec Pnarity 4 NotUsed z Codec Parity 5 NotUsed z Codec Pnonty 6 NotUsed z Codec Pnority 7 NotUsed s Codec Prerity 8 NotUsed x RTP Packet Length G711 amp G729 20 ms 6 723 30 ms z G 723 5 3K G 723 5 3K COn of Voice VAD Voice VAD COn Of It is very important to use only u law or a law as all DTMF is inband So if you wan
18. MV 374 MV 378 VoIP GSM Gateway User Manual PORTech Communications Inc Content BAN TRODUCTION peccet PEUT 1 ZEUNCTION DESCRIPTION a5 pp neo idees 1 PARTS LIST ont nonoui 1 4 DIMENSION z 30X28X4 CM ssl li o 2 SCHART OF THE DEVICE savna ka ana 3 6 WEB PA GE SETTING nail aa aka aka jebe Nam 4 TSXSTEM ENEORM AE ION santa a RANA ea nij 5 O ROUTE uje ONO NE iN NS TUNES 5 D MOBI UE iov NT 12 BORNE WORK nai ea A NA a a ana 20 D SIR LU II M T mess UR TN 25 12 NAT TRANS oT 34 13 SYSTEM AUTHORITY onda ss ges nn nets essaie stat s 35 14 UPDA Essonne lan aie tannins 36 SAVE CHANGE nunnia oa EEUU PNE EEUU RUNI Dre RU Mri CE MR KIM IRE REA VR ER 38 T6 REBOO ise 39 17 dps TEM Ju cse 40 TS SPECTEICA TION pner 42 19 APPENDIX SETUP MV 374 MV 378 WITH ASTERISK ceret eren 43 1 Introduction MV 374 MV 378 is a 4 8 channels VoIP GSM Gateway for call termination VoIP to GSM and origination GSM to VoIP It is SIP based and compatible with Asterisk It can enable to make 4 8 calls simultaneously from IP phones to GSM networks and GSM network to IP phone 2 Function description 2 1 VolP SIP GSM 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
19. Notes 1 Reboot 195 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 120 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 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 111254 IVR will announce the current 40 DNS Server setting in the Primary DNS field Default 192 168 0 1 Check Firmware Version 128 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
20. Status WAN Settings Mask GEM ings MAC 00037 e006666 SNTP Settings SECUTI SIP Settings DHCP Server OOn of _NAT Transform Stene 0 Update End IP 0 System Authority Lease Time lo 0 dd hh Save Change Reboot 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 22 10 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 P CTI vnd 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 Time Zone GMT v 08 00 V hh mm Slave Setting Sync Time 1 0 o dd hh mm SIP Settings NAT Transform Update System Authority Save Change Reboot 23 10 5 Slave Settings Record Slave IP for Master PORTech Interlink Setting ES CTI Adr Route Mobile Master 192 168 0 111 40000 Local i Network
21. TP 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 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 54 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 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 7e45b773130f1fc945efcee502f84042 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt
22. The channel 1 192 168 0 100 5060 The channel 2 192 168 0 100 5062 The channel 3 192 168 0 102 5060 The channel 4 192 168 0 102 5062 The channel 5 192 168 0 104 5060 The channel 6 192 168 0 104 5062 The channel 7 192 168 0 106 5060 The channel 8 192 168 0 106 5062 11 9 Mobile 9 1 Mobile Status PO RTech Mobile Status Your CTI Partner 2008 05 15 17 13 it Mobile 1 v Mobile Status Network Registration Chunghwa Telecom LDM Settings SIM Card ID 144 0 98889602200752095822 Fwd Settings SMS Agent Signal Quality 127 Network GSM S N IMEI 35815600782656 1 E SIP Settings Incoming IP NAT Transform Incoming IP Name Update 3 Outgoing IP System Authority Save Change Incoming Mob Reboot Outgoing Mob 1 Choose Mobile 1 2 3 or 4 MV 378 Mobile 1 2 3 4 5 6 7 8 2 Network Registration The telecom carrier which the SIM card been registered 3 SIM Card ID SIM card ID 4 Signal Quality Signal quality 5 GSM S N IMEI Number 6 Incoming IP The IP address of the last incoming call from LAN 7 Incoming IP Name proxy server name 8 Outgoing IP The IP address of the last outgoing call to LAN 9 Incoming Mob The caller ID of the last incoming call from MOBILE 10 Outgoing Mob The called number of the last outgoing call to MOBILE 12 9 2 Mobile Setting PORTech Your CTI Partner Mobile Setting
23. d SIP Settings NAT Transform Update System Authority Save Change Reboot 35 14 Update In Update you can update the system s firmware to the new one or do the factory reset to let the system back to default setting 14 1 Update firmware PORTech Your CTI Partner Update Firmware You could update the newest firmware PCB mark 2N1494 Route Mobile Method HTTP O TFTP Network SIP Settings NAT Transform SRE Tp nea TER File Location 88 New Firmware TETP Default Settings TFTP Server 192 168 1 250 System Authority Save Change Update Reboot 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 36 14 2 Restore Default Settings In this page Update Default Settings you could restore the factory default settings to the system All setting will restore default setting IP will retain original IP as usual not default IP PO RTech Restore Default Settings Your CTI Partner You could click the restore button to restore the factory settings Route EN Restore
24. default settings Network SIP Settings NAT Transform Update New Firmware De System Authority Save Change Reboot ngg ite ed 37 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 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 Reboot 38 16 Reboot Reboot function you can restart the system If you want to restart the system you can just click the Reboor button then the system will automatically 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 PSyster Authority Save Change Reboot 39 17 IP Setting The operator can setup or query the network parameters by dialing in the mobile number which it SIM card has been put in the main body The status or result is response by voice In the first 20 seconds after power on the 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
25. er please note your asterisk need to have route of destination number 2 specific extension or IP or phone number when mobile call in MV 374 MV 378 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 374 MV 378 will provide dial tone and you can enter mobile number 2 specific mobile number when lan phone call in MV 374 MV 378 will connect with the specific mobile number auto 3 gt It is 1 stage dialing When lan phone and MV 374 MV 378 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 374 MV 378 All changes both need to click save and change 59 60
26. list Please check the parts for any missing parts If do please contact our agents 3 1 MV 374 MV 378 main body 3 2 Power adaptor AC DC 110V AC 12V DC or 220V AC 12V DC 3 3 Network cable 3 4 Antenna MV 374 1 pcs MV 378 2 pcs 3 5 Rackmount compatible with 19 Rack option 3 6 User Manual 3 1 MV 374 3 1 MV 378 3 2 MV 374 3 2 MV 378 SEES 3 4 3 5 option 4 Dimension 30x28x4 cm 5 Chart of the device 5 4 5 5 5 6 5 7 5 8 5 3 Se 5 1 5 1 Antenna Antenna connector 5 2 WAN RJ 45 internet connector standard RJ 45 socket connect to HUB 5 3 DC 12V Power input 5 4 SIM Card 5 5 LINK Indicator Light up when network is connected 5 6 CH3 an indicator light of VoIP3 5 7 CH4 an indicator light of VolP4 5 8 PWR Power LED Light up when power is normal 6 Web Page Setting When the IP setting is done the operator may setup all the rest parameters via web page Browse the IP address from Internet Explorer e g http 192 168 0 100 The following page shows up Login VoIP Enter your username and password to login VolP server Username Password Enter the username and password for authentication default username voip password 1234 The page follows when the username and password are correct 7 System Information 7 1 When you login the web page you can see the demo system current system information like firmware version com
27. n 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 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 374 MV 378 have register proxy server Asterisk The proxy server Asterisk have the route 09 When the callers prefix number is 0932 MV 374 MV 378 will connect 0911123456 automaticlly 2 Mobile to Lan Any caller call the MV 374 MV 378 s sim MV 374 MV 378 will prompt dial tone Caller can enter IP or sip extension or phone 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 8 2 Mobile to LAN Speed Dial Settings When you set M
28. obile to LAN Speed Dial Settings and Mobile to LAN at the same time MV 374 MV 378 will give priority to Mobile to LAN Speed Dial Settings PORTech Mobile To LAN Speed Dial Your CTI Partner Route Mobile To Lan Settings Or m e Mobile To Lan Speed Dial Lan To Mobile Settings 1 Mobile 3 Network 4 SIP Settings 5 NAT Transform 6 Update System Authority Save Change Reboot Add New Phone Position 0 9 Name URL 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 8 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 Your CTI Partner Mobile 1 2 v Route Pa 1 v Mobile To Lan Settings du ECL To Lari ded o uL CallNum O Lan To Mobile Settings Latter m n E Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot M C 4 C l2 O Delete Selected Delete All Add New Position l 0 49 URL Ex 192 168 0 1 192 168 0 ma Ex 0911 25t d 7A77 19t The MV 374 MV 378 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
29. olete transfer mechanism BYE Also Restrict 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 V Remove fancy characters from phone numbers CCE URL Port Fwd to Mobile1 121660100060 Fwd to Mobile2 p meos 7 Fwd to External tS 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 Te 9 4 Mobile SMS Agent PO RTech SMS Agent Eh CTI ai E Mobile 1 2 v Read received SMS Route Mobile Port Status Status Mobile 1 Standby Settings Mobile 2 Not Ready III Rx List Fwd Settings See ces U Bees SMS Agent La MS Sender Via Mobile 1 O2 SIP Settings Does NAT Transform Maximum Number of UCS2 chars for this text box is 70 Update System Authority Message Save Change Reboot You have 70 UCS2 chars remaining for your description 1 Rx List Read received SMS 2 Dest Num the Receiver s phone n
30. pany etc in this page 7 2 Also you can see the function lists in the left side You can use mouse to click the function you want to set up PORTech Mobile VoIP8 o Route Model Name MY 378 Mobile Model Description GSM 900 1800 1900MHz Hans Firmware Version Thu May 15 14 50 55 2008 icc Codec Version Mon Jul 24 10 55 05 2006 SIP Settings NAT Transform Update System Authority 2007 PORTech Communications Inc Save Change Reboot 8 Route Important The route table 50 sets can share by two channels 1 2 ch 3 4 ch 5 6 ch 7 8 ch The setting please refer 9 2 Mobile settin ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for item 25 49 8 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 Mobile 1 2 v Route Page 1 Mobile Network BUS PC NN CT Select oO SIP Settings NAT Transform Update System Authority Save Change Reboot O O JO A RU l0 O Delete Selected Delete All teset Add New Position 0 49 CID Ex 0911111111 0911 URL Ex 192 168 0 1 2St The MV 374 MV 378 will transfer to the URL according to the caller ID of the Mobile CID 1 may enter the whole number e g 0911111111 2 only part of the number prefix e g 0911 means any number starting with 0911 will be accepted 3 means all numbers ca
31. ponses PORTech SIP Responses Setting Your CTI Partner Route E Mobile 486 Busy here Network O 503 Service unavailable RES J Service Domain GON 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 RPort Setting SIP Responses Other Settings NAT Transform Update System Authority H 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 Mail directly For this function 183 must be turn on 32 11 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 11 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 Yo
32. 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 A 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 42 CNG AEC LEC Packet loss 18 5 GSM MV 374 MV 378 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 19 Appendix Setup MV 374 MV 378 with Asterisk 19 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 374 MV 378 lan Asterisk lt internet gt VOIP provider whatever landline To do such a call you just call your MV 374 MV 378 number it has its own simcard then you get an invitation tone then you dial the
33. t 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 You could set the volume of vour phone in this page 2 VolP Volume fio 0 12 VoIP Gain fi 2 0 15 ia LAN DTMF Gain fio 0 12 Mobile In Gain 3 0 4 je Caller ID C Clid amp Fix SIP User iQ Mobile PIN Code On M Code Confirmed These settings seem to be ok just adjust 47 19 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 19 4 Asterisk configuration Once the MV 374 MV 378 is set you have to configure Asterisk On that side you have to setup files as follow 19 5 sip conf GSM VOIP Gateway MV 374 MV 378 103 type friend 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
34. umber 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 Mobile 1 M CERTES Caller ID RE READ 885935385852 08 05 15 15 41 46 18 Click the serial no you can view message as follows SMS Reader Index RemotelD Date Time Z 1 886935386862 08 05 15 15 41 46 MV Serial can send SMS and Receive SMS 19 10 Network In Network you can check the Network status configure the WLAN Settings LAN Setting and SNTP settings 10 1 Network Status You can check the current Network setting in this page PORTech xad Ed CTI Sal Netw OTK Status Route Mobile Type Fixed IP Client IP 192 168 0 110 Network Mask 255 255 255 0 Status Gateway 192 168 0 254 WAN Settings MAC 000372005555 LAN Settings SNTP Settings Ethernet WAN Interface LAN Interface Slave Setting Type Fixed IP Client IP 192 168 0 112 SIP Settings Mask 255 255 255 0 NAT Transform Gateway 192 168 0 254 ere MAC 000372000077 System Authority ACTE TT LAN Interface Save Change Type Fixed IP Client Reboot IP 192 168 0 114 B Mask 255 255 255 0 Gateway 192 168 0 254 MAC 000372000432 Ethernet3 2 WAN Interface LANInterace Type Fixed IP Client Fixed IP Client IP 192 168 0 116 192 168 0 108 Mask 255 255 255 0 255 255 255 0 Gateway 192 168 0 254 192 168 0 254 MAC 00037 000002
35. ur CTI Partner Other Settings Mobile 1 2 v Route Mobile Hold by RFC of Mobile 1 O On of Network Hold by RFC of Mobile 2 O On Off SIP Settings Voice QoS 40 0 63 Service Domain M Port Settings SIP Qos 40 0 63 Codec Settings SIP Expire Time 300 60 86400 sec Codec ID Setting DTMF Setting RPort Setting SIP Response Ot ngs NAT Transform Update System Authority Save Change Reboot 33 12 NAT Transform In NAT Trans you can setup STUN and uPnP function These functions can help your VoIP device working properly behind NAT 12 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 Mobile 1 2 Route Mobile STUN of Mobile 1 Network STUN of Mobile 2 SIP Settings NAT Transform STUN Setting Update System Authority Save Change Reboot STUN Server STUN Port 34 O On of O On Of 478 1024 85535 13 System Authority In System Authority you can change your login name and password PORTech System Authority Your CTI Partner You could change the login username password in this page Route New username Mobile New password Network Confirmed passwor
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