MV-374 / MV-378 VoIP GSM Gateway User Manual PORTech
Contents
1. 31 11 6 2 180 RING OMO es 31 11 6 3 183 Session 31 11 6 4 Dial oic EROR OR EGRE POR Ee ii n 31 12 Transform neee nana B CONES 36 13 System AUtHOFty enero erect ee tete ee sete ee et ete ee es ete see Reed 37 14 Update REPE EROR DOE MEME EMEND E EID 38 15 Save ee 42 16 ReDDDOL uo coii rei ere ceri rrr riri Era o rri eri RE Er ka 43 IAAP Setting EE 44 18 Specificati i insanin dee ERAAN MID AMA IM RET 46 19 Appendix Setup MV 374 MV 378 with 47 20 How to setup Asterisk to receive Caller ID from MV 374 MV 378 54 2 Te SIMPE Ste I C 63 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 VoIP 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 mobile2. 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 13 Set Primary DNS Server 1 15 xxx Must set Static IP first Enter IP address using numbers on the telephone key pad Use the star key when entering decimal point 45 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 46 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 56 850 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 lt gsm network gt MV 374 MV 378 lt lan gt 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 number which is handled by Asterisk If you have some specia
3. Mobile Status Network Registration Chunghwa Telecom LDM Settings SIM Card ID 144 0 98889602200752095822 Fwd Settings SMS Agent Signal Quality 27 Network GSM S N IMEI 35815600782656 1 SIP Settings Incoming IP NAT Transform Incoming IP Name Update 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 Route Mobile Status Settings Fwd Settings SMS Agent Network SIP Settings _NAT Transform Update 0 System Authority Save Change Reboot Only change mobile into on or off just click submit no need to click save change Mobile Setting Mobile 1 2 9 10512 3 LAN Dialtone Gain 9 0 12 1 VoIP Tx Gain Mobile 1 OFF 4 Rout
4. Mobile 1 2 Route Mobile To Lan Setings Mobile Lan Speed Dial Lan To Mobile Settings 1 Mobile a 3 Network 4 SIP Settings 5 NAT Transform 6 Update a System Authority 3 Save Change 2 Reboot Add New Phone Position 0 9 URL The call will be answered and prompt dial tone again When the caller may enter the system will connect the URL as destination E g Num 0 Name test URL 192 168 0 107 When the caller hear dial tone and enter 0 system will connect 192 168 0 107 8 4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE PORTech AN To Mobile Table Your CTI Partner Mobile 1 2 Route 1 v Mobile To Lan Settings T i To Lan Speed C mem TT BS Lan To Mobile Settings C Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot WN OQ Delete Selected Delete All Add New Position 0 49 URL Ex 192 168 0 1 192 168 0 Call Num The MV 374 MV 378 will transfer to the mobile number according to the incoming URL Ex 0911 25t 17 Sd A 15t 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
5. 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 374 MV 378 main body 3 2 Power adaptor AC DC 110V 12V DC or 220V AC 12V DC 3 3 Network cable 3 4 Antenna MV 374 1 pcs MV 378 2 pcs 3 5 Rack mount accessories compatible with 19 Rack option 3 6 User Manual 3 1 MV 374 3 1 378 3 2 374 3 2 378 ee 3 4 3 5 option 4 Dimension 30 28 4 cm 5 Chart of the device 5 9 5 4 5 5 5 6 5 7 5 8 5 3 5 2 5 1 5 1 Antenna Antenna connector 5 2 WAN RJ 45 internet connector gt 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 VolP3 5 7 CH4 an indicator light of VolP4 5 8 PWR Power LED Light up when power is normal 5 9 reboot button reboot ch1 2 without power off 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 VolP Enter your username and password to login VolP server Username Password Enter the username and password for authentication default username voip pass
6. 65
7. 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 Transmitting NAT to 192 168 66 203 5060 SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK672fa67f59c222327 5f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt 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 62 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 e
8. 255 255 0 Update Gateway 192 168 33 254 a MAC 00037 E001 F7F System Authority Save Change Reboot Type Fixed IP Client 27 IP 192 158 33 104 255 255 255 0 Gateway 192 168 33 254 MAC 00037 001 4 Fixed IP Client IP 192 168 33 106 Mask 255 255 255 0 Gateway 192 158 33 254 00037 EDO1FEB 21 10 2 WAN Settings WAN IP Master Default 192 168 0 100 Slaver1 Master 8080 Slaver2 Master 8180 Slaver3 Master 8280 WAN IP Corresponding port 5060 5062 5064 5066 5068 5070 5072 5074 PORTech wan Settings Your CTI Partner You could configure the WAN settings in this page Route ME Client O PPPoE Network Master IP 192 168 0 110 SIP Settings Mask 255 255 255 0 NAT Transform Gateway 1192 158 0 254 Update DNS Server 168 95 192 1 System Authority DNS Serer2 158 95 1 1 Save Change MAC 100037 e001f d Reboot User Name Password 1 The TCP IP Configuration item is to setup the WAN ports 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 you set the Bridge On then the two
9. Fast Ethernet ports will be transparent 4 When you finished the setting please click the Submit button 22 10 3 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 You could set the SNTP servers in this page Route Mobile SNTP Oof Network Primary Server time windows com Status WAN Settings Secondary Server 208 184 49 9 Time Zone GMT w 08 v 00 hh Slave Setting m SIDA Sync Time 1 10 5 0 dd hh mm SIP Settings NAT Transform Update System Authority Save Change Reboot 23 10 4 Slave Settings Record Slave IP for Master PLEASE don t change this page Important PORTech Your CTI Partner Interlink Setting Route DA Master 40000 Local Network Slave 1 9218833102 40000 Status Slave2 192 168 33 104 40000 Slave 3 19216833105 40000 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 Setti
10. Length _NAT Transform G 711 amp G 729 20 ms Update G 723 30 ms System Authority Save Change G 723 5 3K G 723 5 3K O On Of Reboot Voice VAD Voice VAD On Of 27 11 3 Codec ID Setting You can setup the Codec ID in this page Codec ID Setting Your CTI Partner You could set the value of Codec ID in this page Route Codec Type ID Default Value Network 6726 16 ID 23 23 23 6726 24 ID 2 95 255 22 G726 32 ID 2 Service Domain 185 281 2 Port Settings 5726 40 ID 21 85 255 21 a Settings _ RFC 2833 ID 101 gs 255 101 DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot SIP Settings 28 11 4 DTMF Setting You can setup the DTMF Setting in this page PORTech rye Setting Your Partner Route Mobile O 2833 Network Inband DTMF Send DTMF SIP Inf SIP Settings Service Domain Mobile DTMF debounce 80 range 40 200 default 80 step 10ms Part Settings Codec Settings L d RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 29 11 5 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 p
11. Num 1 may enter the whole number e g 0911111111 2 8 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 AZ 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 11 9 Mobile 9 1 Mobile Status PORTech Lor CTI Partner Mobile Status 2008 05 15 17 13 dis Mobile 1
12. of call 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 branch2 z9hG4bK672fa67159c2223275f5ee286d27597a From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 3192 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 046a412f4e7ed4 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 z9hG4bK67 2fa6 7f59c22232 7 5f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 3192 168 66 203 CSeq 11 REGISTER 61 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
13. 192 168 0110 1921680110 1921680110 1921680110 1821680110 19821680110 1921680110 1921680110 08 11 04 15 07 14 34 11 7 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 e Your CTI Partner Other Settings Mobile 1 2 Route Mobile Hold by RFC of Mobile 1 On Network Hold by RFC of Mobile 2 On Off SIP Settings Voice QoS 40 0 63 Service Domain Port Settings QoS 40 0 63 Codec Settings SIP Expire Time 300 60 86400 sec Codec ID Setting Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 35 12 NAT Transform In NAT Trans you can setup STUN and uPnP function These functions can help your VolP 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 yo
14. 60 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 59 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 7e45b773130f1fc945efcee502184042 3192 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 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 3192 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 60 Scheduling destruction
15. 73503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 9192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x 56 Content Length 254 v 0 07 1001 4804366 4807851 IN 192 168 66 145 s X Lite c IN 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 show caller id mobile number 0928492911 This Is X Lite receiving packet Test ok INVITE sip 1002 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 145 7331 rport branch zZ9hG4bK4C4315351FC84CA582D14FB8C25F 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 8aaaaadb5ad53 57 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 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP
16. 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 branch2 z9hG4bK4C4315351F C84CA582D14FB8C25FC3BF 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 58 v 0 o root 2737 2737 IN IP4 192 168 66 202 s session c IN 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 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 branch2 z9hG4bK590e92b551233a10a0ae7 1944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 192 168 66 203 Contact lt sip 1002 192 168 66 203 50
17. JPhone 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 374 MV 378 address 192 168 66 203 5060 username 1002 displayname user_ 1002 test1 pstn gt call 092849291 1 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 55 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 20fa417265e6a26d0b0aae4f551f06f3 9192 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 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 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 6773
18. M to Asterisk it has to be one of these 2 codecs Mobile Setting You could set the volume of vour phone in this page VoIP Volume fio 0 12 VolP Gan fi 2 0 15 i LAN DTMF Gain 0 12 Mobile In Gain iQ Caller ID C Clid Fix SIP User iQ Mobile PIN Code Code Confirmed These settings seem to be ok just adjust 51 19 3 Antenna position Another important thing is to properly place the provided antenna 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 qualify yes disallow all allow ul
19. MV 374 MV 378 VoIP GSM Gateway User Manual PORTech Communications Inc Content T IB ig ets ET CULO gE EE 1 2 F nction description ntm EAE DER SR NEA nenten 1 lISt XX 1 4 Dimension 30x28x4 2 5 Chart of the device v 3 O WebPage SouinG tU EE E 4 T System Inf rmatiO 5 eternet nde tae deta ecole ad leaks 5 8 T Mobile TO ede nba uta go Rabia 6 8 2 Call Back Service 50 sets New feature 8 8 3 Mobile to LAN Speed Dial Settings 9 8 4 LAN to Mobile Settlngs icri EY ect eea es 10 S MObIIG ere pe WN Were Wis we eee 12 S Mobile Status ener oett dns 12 9 2 Mobile Setting iiie e uera bieten etna 13 9 3 Mobile Forward Setting uc ee aes 16 9 4 Mobile SMS Agent zen e sese eu 18 9 5 use AT Command via Telnet or your 20 LOENSEWOEK zoetere S 21 14 SIP Settllig ui ooo crece eel 25 11 6 1 486 busy here 503 Service unavailable
20. URL Port Fwd to Mobile1 pgo Fwd to Mobile2 mienne Fwd to External 0 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 both 5060 port 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 9 4 Mobile SMS Agent PORTech CTI Avira SMS Agent _ 1 2 Read received SMS Route ENIM 888666 Status Mobile 1 Not Ready Rx List Settings Fwd Settings Mobile 2 Not Ready Rx List SMS Agent Network Encode ASCII Fbi ASC7 ASCII 7 bit SIP Settings Va 91 O2 UCS2 Unicode 16 bit NAT Transform Dest Num Update Maximum Number of chars for this text box is 160 System Authority Save Change Message Reboot You have 160 chars remaining for your description end 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 Mobile 1 BEEN C
21. aller ID REC READ 885935385852 08 05 15 15 41 46 18 Click the serial no you can view message as follows SMS Reader Index TT Date Time 1 896935386862 08 05 15 15 41 46 MV Gerial can send SMS and Receive SMS 9 9 5 use Command via Telnet your program Allows your program or Telnet Send receive SMS with AT Command available in PCB194A approximately after April 2008 Telnet PORT Corresponding port as follows Master ip 23 SLAVE 1 8023 SLAVE 1 8123 SLAVE 1 8223 aM cu Please enter account password sx user level 1 and passwor command logout module modulel module ERE 1 Choose module got press ctrl x to release module 1 Enter ate1 then you can see 0 V you at command below at cmgf 1 test gt 20 10 Network In Network you can check the Network status configure the WLAN Settings LAN Setting SNTP settings 10 1 Network Status You can check the current Network setting in this page Network Status Mobile Type Fixed IP Client Fixed IP Client IP 192 168 0 110 192 168 33 254 Network 255 255 255 0 255 255 255 0 1 1 Gateway 192 168 0 254 192 168 33 254 VAN ETT MAC 00037 001 70 00037 EDO1F7E SNTP Settings Slave Setting TEEN NS Type Fixed IP Client SIP Settings IP 192 168 33 102 NAT Transform Mask 255
22. ave Change In Save Change you can save the changes you have done If you want to use new setting in the VolP 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 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 42 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 RTech 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 CSySt ri Authority Save Change Reboot 43 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 Notes 1 Reboot 195 After you hear Option Successful hang up Unit will reboot automatically Factory Reset 196
23. aw prefered codec for DTMF detection allow alaw 19 6 extensions conf 52 kkkkkkkkkk miii 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 374 MV 378 sim card mail box thru GSM exten gt 888 1 exten gt _888 2 Dial SIP EXTEN 103 60 r exten gt 888 3 Hangup 53 20 How to setup Asterisk to receive Caller ID from 374 378 est version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a e X Lite 1105x Modify file the following setting to etc asterisk sip conf 1000 type friend secret 1000 qualify yes nat yes host dynamic canreinvite no context internal 1001 type friend secret 1001 qualify yes nat yes host dynamic canreinvite no context internal 1002 type friend 54 secret 1002 qualify yes nat yes host dynamic canreinvite no context internal Add the following setting to etc asterisk extensions conf internal exten gt 1000 1 Dial SIP 1000 exten gt 1001 1 Dial SIP 1001 exten gt 1002 1 Dial SIP 1002 configure trixbox 2 2 address 192 168 66 202 5060 S
24. e 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 15 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 4 1 Your Pariner Forward Setting Mobile 1 2 Route Mobile Forward Enable Status Settings CS Fwd Settings Fwd to Mobilel E Fwd to Mobile2 Network Fwd to External SIP Settings NAT Transform pe System Authority Save Change Reboot i 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 Se Profile Options General Initialization DTMF Use short headers v Expose software version Use obsolete 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 packetz Voice mail number or address Remove fancy characters from phone numbers
25. elete Selected Delete All reset Add New Position 0 49 6 Ex 0911111111 09117 URL Ex 192 168 0 1 281 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 can be accepted 4 means the calls without the CID Please note the priority of the rules The item which has more digits will have higher priority If the digits are the same then former one gets the higher priority URL The IP address to transfer this call 1 may enter the whole IP address e g 192 168 0 101 or proxy extension or phone number 2 If this field is blank or simply it means refuse to transfer 3Jf 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 wil
26. esidential installation This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or television reception which can be determined by turning the equipment off and on the user is encouraged to try to correct the interference by one or more of the following measures Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver Connect the equipment into an outlet on a circuit different from that to which the receiver 15 connected Consult the dealer or an experienced radio TV technician for help Operation 15 subject to the following two conditions 1 this device may not cause interference and 2 this device must accept any interference including interference that may cause undesired operation of the device 64 FCC RF Radiation Exposure Statement 1 This Transmitter must not be co located or operating in conjunction with any other antenna or transmitter 2 This equipment complies with FCC RF radiation exposure limits set forth for an uncontrolled environment This equipment should be installed and operated with a minimum distance of 20 centimeters between the radiator and your body
27. ing Range 0 to 24 0 49 5 CODEC Tx Gam 6 6 CODEC Rx Gain 8 0 7 7 SIP From Tel User Standard Answer Delay 0 0 15 8 9 CUD Presentation Suppression Invocation 10 Mobile PIN Code On Code Confirmed 1 Answer Mode Answered Alerted Income T Routing Range 25 49 0 49 CODEC Tx Gain e 0 7 CODEC Rx Gain 6 From Tel User Standard Answer Delay 0 15 CLID Presentation Suppression Invocation Mobile PIN Code On O Code Confirmed LAN Answer Mode Answered Alerted Income Mobile 1 1 VoIP Tx Gain 2 VolP Rx Gain Mobile 2 13 1 Tx Gain To adjust the volume of LAN side 2 VolP Rx Gain To adjust the volume of Mobile side 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 5 CODEC Tx Gain as above 6 Rx Gain as above 7 SIP From Caller ID transfer 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 fol
28. l 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 Call Back Service 50 sets PORTech To LAN Table 4 CTI Zend Page cin URL 0833579613 886933579613 Mobile Lan Speed Dial Lan To Mobile Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot KOE Do GOV dx gt Delete Selected Delete All Add New Position 0 49 CID Ex 0911111111 09117 URL 192 168 0 1 251 You can set call back service as the following steps 1 CID set the phone number here up to 50 sets 2 URL is the command of call back Application a Call MV 374 MV 378 b MV 374 MV 378 will detect the phone number is in call back list or not c If yes MV 374 MV 378 will reject the call and call it back d You will receive the call from MV 374 MV 378 and prompt a dial tone 8 3 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time MV 374 MV 378 will give priority to Mobile to LAN Speed Dial Settings PORTech Mobile To LAN Speed Dial Your CTI Partner
29. l 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 47 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 In Bold are the parameters depending on your installation LAN Settings You could configure the LAN settings in this page LAN Mode C Bridge WAN Setting IP Type Mask Gateway DNS Server DNS Server MAC Fixed IP C DHCP Client C PPPoE mw370IP Router IP 48 LAN To Mobile Table Page 2 CN EN 0 your asterisk IP m 1 2 E 3 4 5 7 B E 9 Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM Mobile To LAN Table SS 78 authorised mobile 103 1 anotherauthorised 103 2 3 4 Ez 5 pr 6 F 7 8 9 E The mobile number you give in that page are the authorised mobi
30. le which can call GSM to Asterisk 49 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 C Of Display Name 103 User Name hoa 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 50 Codec Settings You could set the codec settings in this page Codec Priority Codec Pnority 1 6711 dew gt Codec Priority 2 6711 mow Codec Pnarity 3 NotUsed gt Cadec 4 NotUsed 3 Cadec 5 NotUsed 31 Codec Pnonty 6 NotUsed 2 Codec Pnority 7 NotUsed Codec NotUsed gt RTP Packet Length G711 amp G 729 20 ms 6 723 30 ms gt G 723 5 3K G 723 5 3K Of Voice VAD Vaice VAD Of 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 GS
31. lease click the Submit button PORTech RPort Setting Your Partner Mobile 1 2 Route Mobile RPort of Mobile 1 On O Off RPort of Mobile 2 On O Off 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 30 11 6 SIP Responses PORTech Your CTI Partner SIP Responses Setting Route Response port busy Mobile 486 Busy here Network 503 Service unavailable SIP Settings E SIP Responses Service Domain 9ON OOFF 180 Ringing Auto force to if 183 was OFF Codec Settings OON OFF 183 Session Progress Codec ID Setting E E Responses OON Oorr 19216803 Download Dial Peer Other Settings NAT Transform Dial Peer Configuration Table corresponding IP please read next page System Authority Save Ch If you have dial peer server Si server Asterisk set GSM route please set Dial Peer server s IP 11 6 1 486 busy here 503 Service unavailable When Device is busy you can select 486 or 505 to response to SIP 11 6 2 180 Ring on off LAN TO MOBILE two stage dialing can be turn off therefore there will be no the Ring Back Tone all the phone call will be transferred to prompt voice directly For this function 183 must be turn on 11 6 3 183 Session Progress It mean
32. lows 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 Please note If you choose this option please don t register to Asterisk and proxy server Please only fill proxy server 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 lt 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 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 th
33. mark 2N1494 Route Mobile Method HTTP TFTP Network SIP Settings Transform Cade Type 866 Update File Location New Firmware TEE Default Settings TETP 1 MEAE TFTP Sewer 1921681250 System Authority Save Change Reboot Update Reset 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 40 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 will retain original IP as usual not default IP Factory all all setting include ip will restore default setting PORTech Your CTI Partner Restore Default Settings You could click the restore button to restore the factory settings Route Mobile defend Restore default settings Network Restore factory all settings factoryAll included all IP address SIP Settings NAT Transform Update New Firmware Default Settings System Authority Save Change Reboot 41 15 S
34. ng 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 Your CTI Poriner Service Domain Settings Route Mobile 1 Menis Network Active ON O OFF SIP Settings Display 803 User Name 803 Port Settings Register Name 803 cader EE Register Password ese DTMF Setting Domain Server Proxy Server 192 168 0 1 Other Settings Outbound Proxy NAT Transform Status Registered update System Authority Ix O ON OFF Save Change Display Name BEEN 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 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 Pro
35. nter 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 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 63 15 21 Federal Communications Commission FCC Statement You are cautioned that changes or modifications not expressly approved by the part responsible for compliance could void the user s authority to operate the equipment 15 105 b Federal Communications Commission FCC Statement This equipment has been tested and found to comply with the limits for a Class B digital device pursuant to part 15 of the FCC rules These limits are designed to provide reasonable protection against harmful interference in a r
36. s on progressing When you turn 183 on it means you can hear the prompt voice while GSM side is busy We recommend you to turn this on if you use SIP Proxy 11 6 4 Dial Peer Lan to mobile Dial peer software will look for available channel to dial out E g When the first port is busy MV 378 will use the second port to dail out and so forth 31 Edit DialPeer ini Window Xpos 512 Ypos 252 Width 471 Height 399 Info Total 16 VoiplP 1 192 168 0 100 2 192 168 0 100 3 192 168 0 100 4 192 168 0 100 5 192 168 0 100 6 192 168 0 100 7 192 168 0 100 8 192 168 0 100 9 192 168 0 110 10 192 1 112192 1 12 192 1 13 192 1 14 192 1 15 192 1 16 192 1 SipPort 125060 2 5062 3 5064 4 5066 5 5068 6 5070 7 5072 8 5074 9 5060 10 5062 11 5064 12 5066 13 5068 68 0 110 68 0 110 68 0 110 68 0 110 68 0 110 68 0 110 68 0 11 32 14 5070 15 5072 F 16 5074 RtpPort 1260000 2 260002 3 260004 4 60006 5260008 6 260010 7 760012 8 260014 9 260000 10260002 11260004 12260006 13260008 14260010 15260012 16260014 PtcPort 1540000 2 lt 40000 3 40008 4 40008 5 40016 6 40016 7 40024 8 40024 9 40000 10 40000 11 40008 12 40008 13 40016 14 40016 15 40024 16 40024 33 Status lll Dial Peer Oct 29 2008 14 54 07 192 168 0 100 192 168 0 100 192 168 0 100 192 168 0 100 192 168 0 100 192 168 0 100 192 168 0 100 192 168 0 100
37. setting include IP both restore to default setting 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 192 168 0 100 Default Check IP Type 421212 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 11124 IVR will announce the current gateway IP address Default 192 168 0 254 44 Check Primary DNS Server 125 IVR will announce the current setting in the Primary DNS field Default 192 168 0 1 Check Firmware Version 111284 IVR will announce the version of the firmware running Set as DHCP client 521112 The system will change to DHCP Client type 10 Set Static IP Address 1 12 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 1 1 oodt 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 1 14
38. u finished the setting please click the Submit button If you want to set up NAT for MV 374 MV 378 you should install STUN Server first Or it can only allow one way call The initial setting of STUN Server is ON You can download STUN Server here Free www myvoipapp com If you set MV 374 MV 378 at Public IP you can use STUN Servers IP directly 1 1 PO Rech STUN Setting Mobile 1 2 Route Mobile STUN of Mobile 1 On Of Network STUN of Mobile 2 On Of SIP Settings STUN Server stun xten com NAT Transform 6 STUN Port 3478 1024 655535 STUN Setting opas System Authority Save Change Reboot 36 If you set MV 374 MV 378 at Private IP you should use STUN Server s private IP PORTech Route Mobile Network SIP Settings NAT Transform STUN Setting Update System Authority Save Change Reboot STUN Setting Mobile 1 2 On O Off On O Off 192 168 0 90 STUN of Mobile 1 STUN of Mobile 2 STUN Server STUN Port 3478 1024 65535 have installed Dial Peer server please fill Dial Peer server s ip directly 13 System Authority In System Authority you can change your login name and password PORTech nias Route Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot System Authority You could change the login username pass
39. word 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 company 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 VoIPS vooor Model Type 378 Model Description GSM 900 1800 1900MHz M23 Network Firmware Version Tue 4 14 15 35 2008 Codec Version Mon Jul 24 10 55 05 2006 SIP Settings Er North Road Taichung Taiwan NAT Transform Tel 886 4 23058000 Update Fax 886 4 23022596 System Authority E Mail sales portech com tw Web Site http www portech com tw Save Change Reboot 2008 PORTech Communications Inc 8 Route Important The route table 50 sets can share by two channels 1 2 ch 3 4 ch 5 6 ch 7 8 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 To LAN Table Your CTI Partner Mobile 1 2 Route Page 1 Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot CO E CD i d c D
40. word in this page New username New password Confirmed password 37 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 MV 374 have to update 2 times MV 378 have to update 4 times Master IP 8280 first update better gt MV 378 Master IP 8180 MV 378 Master 80802 MV 374 MV 378 Master IP gt MV 374 MV 378 Example Master ip 192 168 0 100 Slaver1 gt 192 168 0 100 8080 Slaver2 gt 192 168 0 100 8180 Slaver3 gt 192 168 0 100 8280 Slaver1 gt 192 168 0 100 8080 Rech Mobile VoIP2 s1 os Model Type MV S1 Model Description GSM 900 1800 1900MHz Firmware Version Thu Aug 7 09 45 25 2008 Codec Version Mon Jul 24 10 55 05 2006 38 Slaver2 gt 192 168 0 100 8180 Rech Mobile VolP2 s2 Model Type MV S2 Model Description GSM 900 1800 1900MHz Firmware Version Thu Aug 7 09 45 25 2008 Codec Version Mon Jul 24 10 55 05 2006 binae Web Management RTech m Mobile VoIP4 s3 os Model Type MV S3 Model Description GSM 900 1800 1900MHz Firmware Version Thu Aug 7 09 45 25 2008 Codec Version Mon Jul 24 10 55 05 2006 39 PORTech ai tht Update Firmware You could update the newest firmware PCB
41. xy 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 Charge Example Register VoipBuster Realm 1 Default Active On C Off Display Name 0922 User jenny0922 Your Voipbuster username Register Name jenny0922 Register Password 7 Your Voipbuster password Domain Server Proxy Server 194 221 562 207 Proxy Server s IP Outbound Proxy Status Registered 26 11 2 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 1 G 711 Network Codec Priority 2 711 alaw SIP Settings Codec Priority 3 G 723 Codec Priority 4 G 729 v Service Domain Codec Priority 5 6 725 18 ef nas Codec Priority 6 G726 24 Codec ID Setting NE DTMF Setting Codec Priority 7 0725 32 RPart Setting Codec Priority 8 G 726 40 SIP Responses Other Settings RTP Packet
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