PORTech DuMV User Manual
Contents
1. DTMF Setting Codec Priority 7 urn qe Yi RPort Setting Codec Priority 8 16 726 40 B SIP Responses NAT Transform G 711 amp G 729 20 ms Update 6 723 30 ms v System Authority J e G 723 5 3K O On Off Reboot Voice VAD On of og VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 4 Codec ID Setting You can setup the Codec ID in this page PORTech Codec ID Setting Your CTI Partner You could set the value of Codec ID in this page Route Mobile Codec Type D DefaultValue Network 6726 16 ID 23 95 256 23 F 6726 24 ID 22 95 255 2 6726 32 ID 2 E 2 Service Domain pone El Port Settings 6726 40 ID 21 95 255 M2 E RFC 2633 ID 40 5 25 101 DTMF setting RPort Setting SIP Responses Other Settings NAT Transform_ Update System Authority _ Save Change Reboot SIP Settings 28 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 5 DTMF Setting You can setup the DTMF Setting in this page PORTech DTMF Setting Your CTI Partner Route Mobile DTMF Transfer to Lan Mobile O 2833 Network O Inband DTMF O Send DTMF SIP Inf SIP Settings 3 Service Domain Mobile DTMF debounce 80 range 40 200 default 80 step 10ms Port Settings l Codec Settings Codec ID Settin DTMF Setting R2. 39 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 CNG AEC LEC Packet loss 18 5 GSM DuMV PCI 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 DuUMV PCI 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 DUMV PCI lt an gt Asterisk lt internet gt VOIP provider lt whatever gt landline To do such a call you just call your DUMV PCI number it has its own simcard then you get an invitation tone then you dial the number which is handled by Asterisk If you have some special deals with your mobile operator like free special number you can call your DUMV PCI for free You can then call all around the world from your mobile at voip cost 19 2 DuUMV PCI 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 DUMV PCI to work with Asterisk you need first to 40 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 configure the box Here are some screen shots showing all the important parameters You have to note that in all the configu
3. 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 48 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Content Type application sdp Content Length 242 v 0 o root 2737 2737 IN IP4 192 168 66 202 s session c IN IP4 192 168 66 202 t 0 0 m audio 15852 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 16 a silenceSupp off SIP 2 0 200 Ok Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt tag 67 7373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f55 1f06f3 192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x Content Length 254 v 0 0 1001 4804366 4807851 IN IP4 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101 a rtpmap 0 pcmu 8000 49 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 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 DuMV g PCI gt hear
4. Example of Application When you call the ch 1 DUMV PCI gsm number it will provide dial tone and you enter a destination number Then ch 2 DUMV PCI will dial this number and connect ch 1 DuMV gPCI mobile to lan set route table ch 2 DUMV PCI lan to mobile set route table Additionally two channels DuMV PCI 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 DUMV PCI The channel 2 DUMV PCI s ip the first ip 5062 e g http 192 168 0 100 5062 252 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 10 Mobile 10 1 Mobile Status PORTech Mobile Status Your CTI Partner 2008 05 16 18 10 R v Mobile 1 Mobile Status Network Registration Chunghwa SENE SIM Card ID BOER Fwd Settings L SMS Agent Signal Quality 17 Network GSM S N acad a SIP Settings Incoming IP NAT Transform Incoming IP Name Update Outgoing IP System Authority Save Change Incoming Mob Reboot Outgoing Mab 1 Network Registration The telecom carrier which the SIM card been registered 2 SIM Card ID SIM card ID 3 Signal Quality Signal quality 4 GSM S N IMEI Number Incoming IP The IP address of the last incoming call from LAN 7 Outgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call fro
5. s 0 63 32 Tel 44 0 1245 808195 Fax 44 0 1245 808299 13 NAT Trans In NAT Trans you can setup STUN and uPnP function These functions can help your VoIP device working properly behind NAT 13 1 STUN Setting you can setup the STUN Enable Disable and STUN Server IP address in this page This function can help your VoIP device working properly behind NAT To change these settings please following your ISP information When you finished the setting please click the Submit button PORTech STUN Setting bbb CTI retrata STUN of Mobile 1 O 0n Off STUN of Mobile 2 O 0n Of Route Mobile Network STUN Server SIP Settings STUN Port 3473 1024 65535 Reset NAT Transform STUN Setting Update System Authority Save Change Reboot 33 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 14 System Auth In System Authority you can change your login name and password PORTech System Authority hb CTI treed You could change the login username password in this page Route New username Mobile y New password Tt Confirmed password l mE B We NAT Transform Update System Authority Save Change Reboot 34 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 15 Save Change In Save Change you can save the changes you have done If you want to use new setting in the VolP s
6. 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 24 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 PORTech Your CTI Partner Service Domain Settings Route Mobile 1 v Network Active O ON O OFF SIP Settings Display Name 3001 ervice Domain Port Settings Register Name 3001 CINE Seas Register Password ense Codec ID Setting DTMF Setting Domain Server Ron Setting Proxy Server 61 218 151 230 SIP Responses gt Other Settings Outbound Proxy NAT Transform Status Not Registered Example Register VoipBuster Realm 1 Default Active On Off Display Name jenny0922 User Name jenny0922 Your Voipbuster username Register Name Register Password Domain Server Proxy Server Outbound Proxy Status jenny0922 ux Your Voipbuster password 194 221 62 207 Proxy Server s IP Reqistered 25 VolPon www voip
7. 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 7e45b773130f1fc945efcee502f840420192 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 55 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 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 DuMV QPCI will provide dial tone and you can enter ip or asterisk extension or phone number If you want to en
8. MOBILE STATUS it eoe cernere ee dte erede een a a Anais 13 102 MOBILE SETTING e eret ree A PER OT IR 14 10 3 MOBILE FORWARD SETTING occcococnnnnonononononononononononononononononnnnnnnnnnnnnnnnnnnnnononononoconononoss 16 10 4 MOBIEEZ SMS AGENT 2 etre dose A ete e etae dala 18 10 5 USE AT COMMAND VIA TELNET OR YOUR PROGRAM oooocccocononnnnnnincconcnnnnonananacricccnnonononos 19 TENE TDT WORK ina ess cue es ecoresesteied eoa Ucesedestesesb ia a teda riari seio ne 20 DZS TR SETTING aiii 24 13 NAT NO TO 33 T4ASYSTEMAUT Hoi iia ee Iud cepa d cuo DE Pa ie nee ca eva EE pa USD Ea a DENM 34 IEEE WAUEOISE LO om 35 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 IGUOPDA TE asi A RT 36 T7IREBOO Viuda tia aia Rai Ricci 38 TS SPECIFICA AAA TT 39 19 APPENDIX SETUP DUMV PCI WITH ASTERISK o ccooncocnconosnoconocnccnnconocanosonoonoss 40 20 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM DUNA OPC leal dial Dodo 46 2L SIMPLE STEPS mr 56 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 1 Introduction DuMV PCI is a 2 channels VoIP GSM Gateway for call termination VoIP to GSM and origination GSM to VoIP It is SIP based and compatible with Asterisk It can enable to make 2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone 2 Function
9. UDP 192 168 66 203 5060 rport branch z9hG4bK6 7 2fa67f59c2223275f5ee286d27597a 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 z9hG4bK67 2fa6 7f59c222327 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 2 192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 12 headers 0 lines Reliably Transmitting NAT to 192 168 66 203 5060 54 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 OPTIONS sip 1002 192 168 66 203 5060 SIP 2 0 Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK7b92dd8a rport From Unknown lt sip Unknown 192 168
10. description 2 1 VoIP SIP GSM DUMV PCI conversion 2 2 50 sets of LAN gt MOBILE routes setting gt 50 sets of MOBILE gt LAN routes setting 2 3 Voice response for setting and status dial in from mobile 2 4 Series connections to save bills 2 5 Standard SIP RFC2543 RFC3261 protocol Communicates with other gateway or PC 3 Parts list Please check the parts for any missing parts If do please contact our agents 3 1 DUMV PCI main body 3 2 Network cable 3 3 Antenna 3 4 User Manual E VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 1 3 4 Dimension 13cm x 32 5cm 2 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 5 Chart of the device 5 1 Antenna Antenna connector 5 2 SIM Slot 2 Insert second SIM card 5 3 SIM Slot 1 Insert first SIM card 5 4 WAN RJ 45 internet connector standard RJ 45 socket connect to HUB 5 5 LAN LAN port It also can be DHCP Server ER VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 6 CABLING 6 1 Connect the internet cable from HUB to the WAN connector of the DuMV PCI If you need to stack up more DUMV PCI you can stack up as follows How to sfack up A LAN WAN em snm A 6 2 Connect the antenna and put it in proper position to get the best signal reception 6 3 Insert the SIM card from
11. which are free to use Profile Options General itializati SIP Proxy Advanced Use short headers Expose software version Use obsolete transfer mechanism BY E Also Restrict caller identity support varies for proxies from T different vendors a Use standard status messages otherwise messages will be taken from SIP packets Voice mail number or address Y Remove fancy characters from phone numbers E a ee URL Port Fwd to Mobile1 poziomo Fwd to Mobile2 meom 0 Fwd to External The Explanation of Picture Fwd to Mobile1 192 168 0 100 5060 it means when 5062 Port are busying SJ Phone can transfer the call to 5060 Port 192 168 0 100 Fwd to Mobile2 192 168 0 100 5062 it means when 5060 Port are busying SJ Phone can transfer the call to 5062 Port 192 168 0 100 e If 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 E VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 10 4 Mobile SMS Agent PORTech sms Agentt Your CTI Partner Read received SMS Route Port Status did Mobile 1 Standby Mobile 2 Standby Status pe TUUS SMS Sender etree DestNum Maximum Number of UCS2 chars for this text box is 70 SIP Settings NAT Transform Message Update Sys
12. 145 CSeq 63148 INVITE User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Type application sdp Content Length 242 v 0 o root 2737 2737 IN IP4 192 168 66 202 s session c IN IP4 192 168 66 202 t 0 0 m audio 13798 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 5 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 a fmtp 101 0 16 a silenceSupp off register issue The packet date from Asterisk as follows Please note user_1002 s display name don t appear So the website s Display Name is not available lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch 29hG4bK590e92b551233a10a0ae71944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 3 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 10 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 3ca93a1e response 4d39ccb0dae64 bb2f1341e9896ac1ea7 uri sip 192 168 66 202 algorithm MD5 User Agent CMI CM5K Content Length 0 11 headers 0 lines Using latest REGISTER request as basis request Sending to 192 168 66 203 5060 NAT Transmittin
13. All Reset Reboot ec 100 US 0 O da DEOS B3 0 C3 The call will be answered and prompt dial tone again When the caller may enter the Num system will connect the URL as destination E g Num 0 Name test URL 192 168 0 107 When the caller hear dial tone and enter 0 system will connect 192 168 0 107 9 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 9 3 Call Back Service 50 sets PORTech Mobile To LAN Table bi CTI nari Route Page 1 cin URL 0933579613 O 886933579613 O Mobile To Lan Speed Dial Lan To Mobile Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot wo Do LOCO GOV dx O NJ CO Delete Selected Delete All Add New Position 0 49 CID Ex 0911111111 0911 URL Ex 192 168 0 1 28t 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 370 b MV 370 will detect the phone number is in call back list or not c If yes MV 370 will reject the call and call it back d You will receive the call from MV 370 and prompt a dial tone 10 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 9 4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE POR
14. DUMV PCI address 192 168 66 203 5060 username 1002 displayname user_1002 47 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 VoIP Web Management Windows Internet Explorer Gc Je rut9 0192 168 66 20308n ogi E x E y pl BRO SEO VAD SSB TAO HAG mene cpuem gt ql so3 e SE SEES G VoIP Web Management fio o Mobile Voip Service Domain Settings You could set information of service domains in this page Route gt Mobile No Mobile 1 y Network Realm 1 Default Active On C Of SIP Settings gt Display Name user 1002 User Ni 1002 NAT Trans M c Register Name 1002 System Auth Register Password Save Change Domain Server 192 168 66 202 Proxy Server 192 168 66 202 Update Outbound Proxy 192 168 66 202 NIRE Status Registered Active On Of Display Name User Nam Register Nami Register Password el E arar 100 7 pstn gt call 092849291 1 mobile number gt DuMV gPCI gt hear the second dial tone call SoftPhone s number gt SoftPhone gt show pstn caller id This Is X Lite receiving packet red word is pstn number Test ok INVITE sip 1001 192 168 66 145 7331 SIP 2 0 Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt Contact lt sip 1002 192 168 66 202 gt
15. DUMVAPCI 2 ports GSM VoIP PCI Card User Manual PORTech Communications Inc VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Content INTRODUCTION ete RIS 1 2 pBUNCTION DESCRIPTION e eaeece cecote icon ua eo cene se aac ce aa sae ese pe dene Seb oe ce alae de ee a eb Oe Deae idad 1 SPAR TS LIS Tinneas fo sPesvkve Gese Deae snas veu casae Velo d las pate OSTA So S OSa Ue Dea des SE UE SERERE 1 4 DIMENSION 13CM X 32 5CM ccccccscscscscssscscccccecscecscececesesesececscesesesesesesesesesesesesesesessseses 2 BS CHART OF THE DEVICE 2 5 eee eoe poo eo aeo ee ecu Poo eo toe ae ea eaa asa oe ases se eda eoe o ad ee Red Ra ei 3 G CABLING unitario ciendo aab el setae Va e Ue ab ob sa EEEE eL oe aene da ae 4 7T WEB PAGE SETTINGQG 5 2 2 aereo ea ea co eoe eee clica 5 8 SYSTEM INFORMAT ION 4 62co eoe ee sea eaae e aeo haee a eae used ee aea ee sao eee e ache oko e a eacus ee eco sake aaa ope dae a pude 6 9 ROUTE sscccccccsciccccdeccccsecseicosnasececsscackcsecaccsecesassiccsaesedeosncacdesscavecocsassbicosoasedesshsacdecacssesacsocsedscees 6 90 1 MOBILE TO LAN SETTINGS ice teet reto deci e ecce tle eee A ect re aveva 6 9 2 MOBILE TO LAN SPEED DIAL SETTINGS coooccccccnonononnnenocncnnnnononenenococnnononnnnnnanoconnnnononanansos 8 9 3 CACI BACK SERVICE OU SETS cc la date ais 10 9 4 EAN TOMOBIEE SETTINGS eii edet tee eee tete a alada 11 IKRILO TIE ge 13 10
16. P 12 7 2 180 Ring on off LAN TO MOBILE two stage dialing can be turn off therefore there will be no the Ring Back Tone all the phone call will be transferred to Prompt voice directly For this function 183 must be turn on 12 7 3 183 Session Progress gt It means on progressing When you turn 183 on it means you can hear voicemail while GMS side is busy We recommend you to turn this on if you use SIP Proxy 31 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 8 Other Settings Other Settings you can setup the Hold by RFC and QoS in this page To change these settings please following your ISP information When you finished the setting please click the Submit button The QoS setting is to set the voice packets priority If you set the value higher than 0 then the voice packets will get the higher priority to the Internet But the QoS function still need to cooperate with the others Internet devices PO RTech Other Settings Lir CTI ceri Route Hold by RFC of Mobile 1 O On Off Hold by RFC of Mobile 2 O On of Mobile Network Voice QoS 40 0 63 SIP Settings SIP QoS SIP Expire Time e 300 60 86400 sec Service Domain Port Settings Codec Settings Cadec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update _ System Authority Save Change Reboot VolPon www voipon co uk sales voipon co uk
17. Port Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 29 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 6 RPort Function You can setup the RPort Enable Disable in this page To change this setting please following your ISP information When you finished the setting please click the Submit button 1 PORTech RPort Setting Your CTI Partner Route RPort of Mobile1 9 On O Of Mobile RPort of Mobile 2 90n O Off Network Gina Reset SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 30 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 7 SIP Responses PORTech Your CTI Partner SIP Responses Setting Route Response on port busy Mobile 486 Busy here Network O 503 Service unavailable _ SIP Settings SIP Responses Service Domain SON OOFF 180 Ringing Auto force to ON if 183 was OFF Port Settings OON 90FF 183 Session Progress Codec Settings Codec ID Setting DTMF Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 12 7 1 486 busy here 503 Service unavailable When Device is busy you can select 486 or 505 to response to SI
18. Tech LAN To Mobile Table Ue CTI iae Route Page Y ves TolarSctros a TET a pa To Mobi Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot cS 00 EA ICE EA he DES E A ES Delete Selected Delete All The DuMV QPCI will transfer to the mobile number according to the incoming URL URL The IP address of the incoming call may enter the whole IP address e g 192 168 0 101 or proxy server s extension If a simple is entered means no restriction for the incoming IP address 11 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 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 A 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 DUMV PCI 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 DUMV PCI will connect this call auto
19. ain DTMF Reciver is not good you can adjust gain down 4 ON Off If you use this channel please click on Otherwise please click off 5 CODEC Tx Gain as above 6 CODEC Rx Gain as above 7 SIP From Caller ID transfer e Tel User Standard If you need to register to Asterisk and proxy server please choose this option And how to transfer the caller ID to LAN please refer 21 How to setup Asterisk to receive Caller ID from DUMV PCI page 42 DuMV PCI will send the message as follows in the Packet From caller number lt sip 3001 192 168 0 228 gt tag 51088abb e Tel Tel DuMV PCI will send the message as follows in the Packet From caller number lt sip caller number 192 168 0 228 gt tag 6ac93f7c Please note lf you choose this option please don t register to Asterisk and proxy server Please only fill proxy server ip and choose Active on else field empty in sip setting service demain e User Tel DuMV PCI will send the message as follows in the Packet From Username sip caller number 192 168 0 228 gt tag 7f130947 lt If you choose this option please don t register to Asterisk and proxy server Please only fill proxy server ip Username and choose Active on else field empty in sip setting service demain 15 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 8 Presentation CLIR If you need to bl
20. ales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 16 2 Restore Default Settings Default Setting you can restore the system to factory default in this page You can just click the Restore button then the system will restore to default and automatically restart again PORTech Restore Default Settings ha CTI adit You could click the restore button to restore the factory settings Route Ww Restore default settings Network SIP Settings NAT Transform Update New Firmware Default Settings System Authority Save Change Reboot 37 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 17 Reboot Reboot function you can restart the system If you want to restart the system you can just click the Reboor button then the system will automatically PO RTech Reboot System Eo CTI eta You could press the reboot button to restart the system Route Mobile Reboot system Network SIP Settings NAT Transform Update System Authority Save Change 38 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 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
21. back of the main body take the slide off first 6 4 Connect the power adaptor The POWER LED should be light up 4 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 7 Web Page Setting When the IP setting is done the operator may setup all the rest parameters via web page Browse the IP address from Internet Explorer e g http 192 168 0 100 The following page shows up Login PORTech VoIP Enter your username and password to login VoIP server Username Password TE Remember last login Enter the username and password for authentication default username voip password 1234 The page follows when the username and password are correct 5 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 8 System Information 8 1 When you login the web page you can see the demo system current system information like firmware version company etc in this page 8 2 Also you can see the function lists in the left side You can use mouse to click the function you want to set up PORTech D reel Mobile VoIP2 501a Route Model Name MV 372 Mobile Model Description GSM 900 1800MHz Mao Firmware Version Fri May 16 11 30 35 2008 Codec Version Mon Jul 24 10 55 05 2006 SIP Settings NAT Transform Update System Autharity 2007 PORTech Communications Inc Save Change Reboot 9 Route 9 1 Mobile TO LAN Se
22. ed 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 DuMV PCI have register proxy server Asterisk The proxy server Asterisk have the route 09 When the caller s prefix number is 0932 DuMV Q PCI will connect 0911123456 automaticlly 2 Mobile to Lan Any caller call the DuMV QPCI s sim DUMV PCI will prompt dial tone Caller can enter IP or sip extension or phone number sip extension or phone number both need to register SIP Proxy Server or Asterisk Phone number SIP Proxy Server or Asterisk need to set the route of this phone number 9 2 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time DUMV PCI will give priority to Mobile to LAN Speed Dial Settings 8 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 PORTech Mobile To LAN Speed Dial Your CTI Partner Route MA O E7773 Test 192 168 0 107 O Mobile To Lan Settings Mobile To Lan Speed Dial O MUD L Mobile SIP Settings NAT Transform Update System Authority Save Change Delete Selected Delete
23. etting in this page PORTech Network Status Your CTI Partner tno Mobile Type Fixed IP Client Fixed IP Client IP 192 168 0 122 192 168 0 102 Network Mask 255 255 255 255 255 255 0 Gateway 192 168 0 254 192 168 0 254 WAN Settings MAC 00037E009999 00037 E008868 LAN Settings SNTF Settings SIP Settings NAT Transform Update System Authority Save Change Reboot 20 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 11 2 WAN Settings You can check the current Network setting in this page 1 The TCP IP Configuration item is to setup the WAN port s network environment You may refer to your current network environment to configure the system properly 2 The PPPoE Configuration item is to setup the PPPoE Username and Password If you have the PPPoE account from your Service Provider please input the Username and the Password correctly 3 The Bridge Item is to setuo the system Bridge mode Enable Disable If you set the Bridge On then the two Fast Ethernet ports will be transparent 4 When you finished the setting please click the Submit button PU RTech WAN Settings Your CTI Partner You could configure the WAN settings in this page Route Network Mode O Bridge 9 NAT Mobile Network WAN Setting IP Type O Fixed IP O DHCP Client O PPPoE otatus T gt WAN Settings IP 192 168 0 122 LAN Settings Mask 255 255 255 0 SNTP Setti
24. g 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 59 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 eived 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502184042 3 192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 Transmitting NAT to 192 168 66 203 5060 SIP 2 0 401 Unauthorized 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 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY WWW Authenticate Digest algorithm MD5 realm asterisk nonce 5def9231 Content Length 0 Scheduling destruction of call 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 53 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Via SIP 2 0
25. l 44 0 1245 808195 Fax 44 0 1245 808299 Service Domain Settings Realm 1 Default Active ON O OFF Display Name 1103 User Name 103 Register Name 1103 Register Password Domain Server Asterisk IP Proxy Server Outbound Proxy Status Not Registered Once Asterisk configuration is made you should get Registered on the Realm1 Codec Settings Codec Priority Codec Priority 1 G 711 u law Codec Priority 2 G 711 a law Codec Priority 3 Mot Used Codec Priority 4 Not Used Codec Priority 5 Not Used Codec Priority 6 Not Used Codec Priority 7 Not Used Codec Priority 8 Not Used G 711 amp G 729 20 ms 6 723 30 ms Y G 723 5 3K O On Off Voice VAD O On Off 43 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 It is very important to use only u law or a law as all DTMF is inband So if you want to be able to do some DISA when you call from GSM to Asterisk it has to be one of these 2 codecs Mobile Setting WolP Tx Gain 10 0 12 VoIP Rx Gain 3 0 15 LAN Dialtone Gain 10 0 12 Mobile ON OFF Routing Range 0 lio 49 0 49 CODEC Tx Gain le p 7 CODEC Rx Gain le D 7 SIP From Tel User Standard Answer Delay lo 0 15 CLID Presentation Suppression Invocation These settings seem to be ok jus
26. m MOBILE 2 3 4 5 6 Incoming IP Name proxy server name 7 8 9 Outgoing Mob The called number of the last outgoing call to MOBILE 13 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 10 2 Mobile Setting PORTech Mobile Setting Your CTI Partner Route 1 VoIP Tx Gain lg 0 12 2 SIRRI Gain 11 9 15 Mobile G LAN Diattone Gain 9 12 peg Fwd Settings 4 Routing Range lo to 49 D 49 SMS Agent 5 CODEC Tx Gain e p 7 6 CODECRxGain e 0 7 Network 7 SIP From Tel User Standard Y Answer Delay D 0 15 12 SIP Settings 8 CLID Presentation Suppression Invocation NAT Transform 9 Mobile PIN Code On Code Confirmed Update LAN Answer Mode Answered Alerted O Income System Authority 10 Save Change Reboot Routing Range lo to 49 0 49 CODEC Tx Gain 6 0 7 CODEC Rx Gain 6 0 7 SIP From Tel User Standard M Answer Delay D 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 Mobile 2 Rx EE 2 VoIP Rx Gain 1 VoIP Tx Gain To adjust the volume of LAN side 14 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 2 VoIP Rx Gain To adjust the volume of Mobile side 3 LAN Dialtone G
27. n PORTech SNTP Settings Your CTI Partner You could set the SNTP servers in this page Route Mobile SNTP On Oof Network Primary Server itime windows com Status P EE zi WAN Settings Secondary Server 208 184 49 9 LAN Settings SNTP Settings Time Zone GMT g Y 08 vi 00 xi hh mm SIP Settings Sync Time a hl hb dd hhimm NAT Transform Update System Authority Save Change Reboot 23 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 SIP Setting In SIP Setting you can setup the Service Domain Port Settings Codec Settings RTP setting RPort Setting and Other SettingS If the VoIP service is provided by ISP you need to setup the related informations correctly then you can register to SIP Proxy Server correctly 12 1 In Servcie Domain Function you need to input the account and the related informations in this page please refer to your ISP Provider You can register three SIP accounts You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from the tree SIP account First you need to click Active to enable the Service Domain then you can input the following items 1 No choose Mobile 1 or Mobile 2 2 Display name you can input the name you want to display 3 User name you need to input the User Name get from your ISP 4 Register Name you need to input the Register Name get from your ISP
28. ngs PNP AES Eu ido Gateway 192 168 0 254 SiPSetings DNS Server 168 95 192 1 NAT Transform DNS Server2 168 95 1 1 Update MAC 00037 e009999 System Authority Save Change PPPoE Setting Reboot User Name Password Me VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 11 3 LAN Settings You can check the current Network setting in this page 1 The TCP IP Configuration item is to setup the WAN port s network environment You may refer to your current network environment to configure the system properly 2 DHCP Server You may refer to your current network environment to configure the system properly PO RTech LAN Settings Your CTI Partner Route IP 1192 168 0 102 Mobile i Mask 255 255 255 0 _Network MAC 00037 8008888 Status Co LAN Settings ettings DHCP S SNTP Settings MSS L2 sa b 2d Start IP 450 SIP Settings a End IP 200 NAT Transform t h 3 i ease Time dd hh Update 5 System Authority 7 Save Change 0 Reboot zo VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 11 4 SNTP Settings SNTP Setting function you can setup the primary and second SNTP Server IP Address to get the date time information Also you can base on your location to set the Time Zone and how long need to synchronize again When you finished the setting please click the Submit butto
29. ock the Caller Id for call termination please choose Suppression 9 Mobile PIN Code If you need to unlock pin code via DUMV PCIl you can click On and enter pin code 10 LAN Answer Mode Answered when mobile answer then connect the call Alerted when the mobile is ringing back tone then connect the call Income when lan dial out then connect soon 11 Band Type When you buy Quad band you need to choose your GSM frequency 12 Answer Delay Delay for incoming call when the ring 10 3 Mobile Forward Setting When the first route are busying SIP can transfer phone call to another free route When the device are busying the phone call can be transfer to another device external equipments PORTech Forward Setting Your CTI Partner Route O Forward Enable Mobile a AA URL Port Fwd to Mobilet 192 168 0 100 5060 Fwd to Mobile2 192 168 0 100 5062 Status Settings tings Fwd Settings SMS Agent Fwd to External JN Network SIP Settings NAT Transform Update System Authority Save Change Reboot ii Forward Enable is not motivate on Defualt value 16 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 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
30. on co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 2 Port Setting You can setup the SIP and RTP port number in this page Each ISP provider will have different SIP RTPport setting please refer to the ISP to setup the port number correctly When you finished the setting please click the Submit button PORTech Ports Setting Your CTI Partner Route Port of Mobile 1 Mobile SIP Port 5060 102465535 Network RTP Port 60000 1024 65535 SIP Settings Port of Mobile 2 Service Domain SIP Port 5062 1024 65535 Port Settings eooo tting RTP Port 60100 1024 65535 Ode P 5 Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update E System Authority Save Change Reboot 26 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 12 3 Codec Settings You can setup the Codec priority RTP packet length in this page You need to follow the ISP suggestion to setup these items When you finished the setting please click the Submit button PORTech Codec Settings Es CTI eost Route Mobile Codec Priority 1 G 711 u law v Network Codec Priority 2 6 711 alaw y SIP Settings Codec Priority 3 Grea 1 Mj Codec Priority 4 6 729 v Service Domain Port Settings Codec Priority G 726 16 wj Codec Settings Codec Priority 6 G7268 24 w Codec ID Setting
31. ration process the DUMV PCI is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation WAN Settings You could configure the WWAN settings in this page WAN Setting IP Type O Fixed IP CO DHCP Client O PPPoE IP Mv370 IP Mask 255 255 255 0 Gateway Router IP DNS Serverl 168 95 192 1 DNS Server2 168 95 1 1 MAC PPPoE Setting User Name Password Submit J Reset Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM LAN To Mobile Table Page 1 HUT A ETA STA O Your Asterisk IP m 1 2 3 4 Ey 5 7 d S 9 41 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Mobile To LAN Table Page 1 Y Mem UR Select 0 Authorised Mobile 103 1 Another Authorised Mobile 103 2 2 3 4 5 6 7 8 9 The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk These mobile number must be defined as your GSM provider displays the number If you don t know how it is displayed just give a call to the box and check the number given in the Incoming Mob field of the Mobile Status page Any number which is not in that list won t have acces to the LAN side so to Asterisk If you want to allow any number just set in that field but beware of the bill 42 VolPon www voipon co uk sales voipon co uk Te
32. second dial tone and call pstn gt pstn answer gt 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 z9hG4bK4C4315351FC84CA582D14FB8C25F C3BF From user_1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt Contact lt sip 1001 192 168 66 145 7331 gt Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE Proxy Authorization Digest username 1001 realm asterisk nonce 0d3b2879 response 8aaaaa5b5ad53 654bf0a2ab0fa9bb1 18 uri sip 1002 192 168 66 202 algorithm MD5 Max Forwards 70 Content Type application sdp User Agent X Lite release 1105x Content Length 254 v 0 o 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite 50 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101 a rtpmap 0 pcmu 8000 a rtpmap 8 pcma 8000 a rtpmap 3 gsm 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 145 7331 branch z9hG4bK4C4315351FC84CA582D14FB8C25FC3BF sreceived 192 168 66 145 rport 7331 From user_1001 lt sip 1001 192 168 66 202 7331 gt tag 1 121869743 To lt sip 1002 192 168 66 202 gt tag as2a2fbf98 Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66
33. t 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 DUMV PCI 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 45 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 20 How to setup Asterisk to receive Caller ID from DuMV PCI Test version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a e X Lite 1105x Modify file e Add the following setting to etc asterisk sip conf 1000 type friend secret 1000 qualify yes nat yes host dynamic canreinvite no context internal 1001 type friend secret 1001 qualify yes nat yes host dynamic canreinvite no context internal 1002 type friend secret 1002 qualify yes VolPon www voipon co uk sales voipon co uk 46 Tel 44 0 1245 808195 Fax 44 0 1245 808299 nat yes host dynamic canreinvite no context internal e Add the following setting to etc asterisk extensions conf internal exten gt 1000 1 Dial SIP 1000 exten gt 1001 1 Dial SIP 1001 exten gt 1002 1 Dial SIP 1002 configure trixbox 2 2 address 192 168 66 202 5060 SJPhone address 192 168 66 145 5060 username 1000 displayname user_1000 X Lite address 192 168 66 145 7331 username 1001 displayname user_1001
34. t adjust 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 DUMV PCI is set you have to configure Asterisk On that side you have to setup files as follow 19 5 sip conf GSM VOIP Gateway DUMV PCI 103 type friend 44 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 username 103 fromuser 103 regexten 103 When they register create extension 401 secret xxxxxxx Asterisk extension password context gateway Incoming calls context dtmfmode inband Very important for DISA to work call limit 1 Limit to 1 call max callerid GSM Gateway lt 103 gt host dynamic nat no Gateway is not behind a NAT router canreinvite no Typically set to NO if behind NAT insecure very qualify yes disallow all allow ulaw prefered codec for DTMF detection allow alaw 19 6 extensions conf eii 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 g
35. tem Authority _ You have 70 UCS2 chars remaining for your description Save Change Reboot 1 Rx List Read received SMS 2 Dest Num the Receiver s phone number 3 Message Please fill the message that want to send to receiver When you click Rx List you can view all received SMS as follows SMS Rx List Read Status RemotelD 3 REC READ 006935114545 08 01 01 19 34 22 2 REC READ 885935385852 08 03 12 16 25 27 18 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Click the serial no you can view message as follows SMS Reader TAERE 886935386862 08 03 12 16 25 27 Mw Serial can send SMS and receive SMS Back Delete 10 5 use AT Command via Telnet or your program Allows your program or Telnet Send receive SMS with AT Command Port 23 username volp Please enter account password user level 1 and password module modulel module 2 command logout gt modulel getting module 1 Choose module got press ctrl x to release module 1 q 1 Enter ate1 then you can see 0 your at command below at cmgf 1 gt test gt 19 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 11 Network In Network you can check the Network status configure the WLAN Settings LAN Setting and SNTP settings 11 1 Network Status You can check the current Network s
36. ter 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 DUMV PCI 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 DUMV PCI will provide dial tone and you can enter mobile number 56 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 2 specific mobile number when lan phone call in DUMV PCI will connect with the specific mobile number auto 3 gt It is 1 stage dialing When lan phone and DuUMV PCI 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 DUMV PCI All changes both need to click save and change 57 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299
37. ttings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN 6 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 PORTech mobile To LAN Table Your CTI Partner Route Page Y A Select Mobile To Lan Speed Dipl E Lan To Mobile Settings Mobile 2 Network 3 SIP Settings i 5 NAT Transform a Update 7 System Authority 8 Save Change 9 Reboot Delete Selected Delete All reset Add New Position 0 49 CID Ex i911111 071 209117 URL Ex 192 168 0 1 2St reset The DUMV PCI 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 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 Te VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 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 3Jf an entered it means 2 stages dialing The call will be answer
38. ystem You have to click the Save button After you click the Save button the system will automatically restart and the new setting will effect PORTech Mb CTI abil save Changes You have to save changes to effect them Route l Mobile i Network SIP Settings NAT Transform Update New Firmware Default settings System Authority Save Change Reboot VolPon www voipon co uk sales voipon co uk save Changes 35 Tel 44 0 1245 808195 Fax 44 0 1245 808299 16 Update In Update you can update the system s firmware to the new one or do the factory reset to let the system back to default setting 16 1 Update firmware 1 In New Firmware function you can update new firmware via HTTP in this page You can upgrade the firmware by the following steps 2 Select the firmware code type Risc code 3 Click the Browse button in the right side of the File Location or you can type the correct path and the filename in File Location blank 4 Select the correct file you want to download to the system then click the Update button PORTech A ah dea Update Firmware You could update the newest firmware PCB mark 2K123B Route Mobile Method 10 HTTP O TFTP Network SIP Settings NAT Transform Code Type Risc v NP j Update File Location Je au ettings TFTP Sewer 192 158 1 250 System Authority Save Change Reboot 36 VolPon www voipon co uk s
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