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1. 0 49 URL Ex 192 168 0 1 192 168 0 Call i Ex 0911 2St d d A 1St Num Add Reset When the Lan of the Mobile VoIP is called this device transfers the call to Call Num according to the URL of the incoming call 8 3 1 URL The IP address or proxy extension numbers of the incoming call You could set the URL as the following formats 1 The complete IP address e g 192 168 0 101 2 The proxy extension numbers e g 103 3 Part of an IP address plus e g 192 168 0 This means the IP 10 address starting with 192 168 0 would be accepted to transfer 4 Part of the proxy extension numbers plus e g 10 This means the extension numbers starting with 10 would be accepted to transfer 8 3 2 Call Num the phone numbers of destination You could set the Call Num as the following formats 1 The complete number e g 0911111111 2 this means to transfer via 2 stage dialing The call will be answered with a prompt dial tone for the caller to press the destination phone numbers e g 0911111111 3 F this allow the caller with lan phone dial directly the destination numbers Precondition 1 Mobile VolP and incoming lan Phone are both registered at proxy server or Asterisk 2 Proxy server asterisk has set the routing rules to assign specific prefix of numbers to be transferred from Mobile VolP 3 Lan to Mobile routing sets Usage You could dial on your lan phone call any destination
2. 19 Applications 1 Connect to VoipBuster a Register VoipBuster account at Service Domain b Route setting Mobile to Lan set When you call in GSM number of Mobile VolP you can enter destination number that will dial out from VoipBuster Landline is free GSM rate is cheap 2 How to apply 2 sets of Mobile VolP 1 When you call the no 1 Mobile VoIP gsm number it will provide dial tone and you enter a destination number Then no 2 Mobile VolP will dial this number and connect Step 1 n0 1 Mobile VolP mobile to lan set route table Step 2 no 2 Mobile VolP lan to mobile set route table Step 3 Additionally two pcs Mobile VolP both need to register proxy server Step 4 And proxy server set the route that the prefix of destination number to dial out from no 2 Mobile VoIP 2 When you call the no 1 Mobile VoIP gsm number no 2 Mobile VolP will dial this specific number and connect Step 1 no 1 Mobile VoIP mobile to lan set route table specific destination number Step 2 n0 2 Mobile VolP lan to mobile set route table Step 3 Additionally two pcs Mobile VolP both need to register proxy server Step 4 And proxy server set the route that the prefix of destination number to dial out from no 2 Mobile VoIP _4 20 Simple Steps Step 1 Change the Network setting if you need Network network setting Step 2 Register SIP proxy Server or Asterisk or VoipBuster if you need SIP setting service domain Ste
3. Transbox ser Manual Content i INFRODUCTION is 1 2 EFUNGIIONS mannaia 1 3 THE CONTENTS IN PACKAGE lina 2 4 DIMENSION AND PANEL DESCRIPTION coocccccccccocconcocoocococccccoconoccccnnocnccccccncnccccocoooooss 3 SAC CESSORY ATTACHVMEN E sccssrscscssiascscssinsoseasoiascovsensasasvotsusaseeson sassossvenssensnsaevessusosecees 3 6 SETTING AND MANAGING VIA WEB PAGE sssssssesscccceccccccccssssssescecccseeeecoooossssssssee 4 1S5YSTEM INFORMA TTO Nonono EAEE 5 A E 6 MOBILE RALE 12 UM MAA o REIT E 19 PD SER SETTING nodos ra iti 23 IZ NAT TRANS APART PU E ER E E E 32 13 SYSTEM AA E O 32 II SAVE CHANG E sisi RETE NERI EEES eE INNER SLI RS 34 ISUPDA AA rara anatre arie 35 LOREBOOT cziiga nei aaa ARA AS 37 17 SETTING AND CHECKING VIA IVR iisarne aria 38 ES OPE CIBRICA TION AAA Po Un mu A 40 15 ARP EEC ATION Sidonie Io concede arri 41 20 SIMPLESTERS albania 21 APPENDIX SETUP TB 2001 WITH ASTERISK 22 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM 1 Introduction TB 2001 series products provide you the best connect solution for heterogeneous network including WLAN GSM or PSTN You may use a SIP protocol VolP phone or software to connect to the TB 2001 then reach this call to the mobile network and vice versa With multiple sets of TB 2001 you may even build an international call network 2 Functions 2 1 VolP SIP GSM conversion 2 2 VolP SIP CDMA conversion 2 3 Voice
4. 192 168 1 1 J Se Outbound Proxy 192 168 11 RPort Setting utboun roxy i gi SIP Responses Status Registered Other Settings NAT Transform Active C ON OFF Update Display Name System Authority Save Chanae User Name Reboot Register Name 1 Active click On to enable the function in Service Domain then input the following items 2 Display name input the name you would like to display 3 User name input your user name in ISP DI 4 Register Name input your register name in ISP 5 Register Password input your password in ISP 6 Domain Server input the Domain Server IP address 7 Proxy Server input the Proxy Server IP address 8 Outbound Proxy input the Outbound Proxy IP address If your ISP does not provide the information you could skip this item 9 After setting click the Submit button Remember to click Save Charge 10 You can see the Register Status in the Status item Example Register VoipBuster Active On Off Display Name fenny0922 User Name fenny0922 Your Voipbuster username Register Name fenny0922 Register Password pp Your Voipbuster password Domain Server o Proxy Server 1194 221 62 207 Proxy Servers IP Qutbound Proxy PO Status Registered 24 11 2 Port Setting You can setup the SIP and RTP port number in this page Each ISP provider will have different SIP RTP port setting please refer to the ISP to setup the port
5. 255 255 255 0 Gateway 192 169 1 254 MAC 00037E002A0C NAT Transform Update System Authority Save Change Reboot 10 2 Network Network Settings VAN Settings You can check the current Network setting in this page The default IP is 192 168 0 100 you could change it to any available IP address or select different IP type to suit your environment 20 TAL WAN Settings You could configure the WAN settings in this page Route Network Mode Bridge NAT Mobile Ss WAN Setting IP Type Fixed IP DHCP Client PPPoE Status Master IP 192 168 1 13 Mask 255 255 255 0 SIP Settings Gateway 1192 168 1 254 NAT Transform DNS Server 205 152 144 23 Update DNS Server2 0 0 0 0 System Authority MAC 00037e002a0c Save Change Reboot PPPoE Setting User Name Password Submit Reset 1 LAN Mode select NAT 2 Fixed IP 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 3 DHCP client you could refer to your current network environment to configure the system properly 4 PPPoE If you have the PPPoE account from your Service Provider please input the Username and the Password correctly 5 After you input or modify the value click the Submit button 10 3 Network SNTP Settings 21 You could select On to give SNTP fu
6. Please note the priority of the routing rules the CID with more digits gets the priority 8 1 2 URL The IP address of destination You could set the URL as the following formats 1 The complete IP address e g 192 168 0 101 2 The proxy extension numbers 3 The phone numbers Note If the device has registered at proxy server Asterisk you can enter any destination phone number Also note that in the proxy server Asterisk you need to set the route of destination phone number 4 Leave it blank or N this mean to refuse to transfer 5 this means to transfer via 2 stage dialing The call will be answered with a prompt dial tone for the caller to press the IP address proxy extension or any phone number as destination The caller press the IP address on the phone keys 192 168 0 101 as 192 168 0 101 8 1 3 Example of Mobile to Lan setting 1 Mobile to Lan 0932 0911123456 When the GSM numbers of the device is called if the caller s prefix numbers are 0932 Mobile VoIP transfers the call to 0911123456 then 0911123456 rings while available Precondition a Mobile VolP has registered at proxy server Asterisk b The proxy server Asterisk has the route of 09 2 Mobile to Lan Any incoming call gets a prompt dial tone so the caller can enter any IP address sip extension or phone number Precondition a SIP extension or phone number needs to register at SIP Proxy Server or Asterisk b Phone
7. Type II _ Default Value TEP T G726 16 ID 23 95 255 M 23 SIP Settings G726 24 ID 22 95 255 lv 22 G726 32 ID 2 Iv 2 Service Domain eee Port Settings G726 40 ID 1 95 255 Y 21 Codec Settings RFC 2833 ID hoi 95 255 Vv 101 DTMF Setting RPort Setting Submit Reset SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot DI 11 5 DTMF Setting You can setup the DTMF Setting in this page TAL DIMF Setting Route AS Mobile DIMF Transfer to Lan i 2833 Network C Inband DTMF SIP Settings C Send DTMF SIP Info Service Domain Mobile DTMF debounce lso range 40 320 default 80 step 10ms Port Settings Codec Settings Codec ID Setting Submit Reset SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot Note If this device has registered at SIP Proxy Server Asterisk please select 2833 If not please select Inband DTMF 28 11 6 RPort Setting You can setup the RPort Enable Disable according to your ISP information After setting remember to click the Submit button LL RPort Setting outs RPort of Mobile On Off Mobile Network Submit Reset SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting Settna CAN c SIP Responses Other Settings NAT Transform Up
8. incoming call when the ring 9 3 Mobile Forward Setting When the first route are busying SIP can transfer phone call to another free route When the device are busying the phone call can be transfer to another device external equipments LL Forward Setting peer I Forward Enable Mobile Name URL Port Status Fwd to External Settings submit cancel Network SIP Settings NAT Transform Update 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 16 redirection replies Turn on the Forward Enable therefore the SJ Phone can designate a port which are free to use Sue Profile Options General Initialization SIP Proxy Advanced DTMF STUN M Use short headers Y Expose software version Use obsolete transfer mechanism BYE Also n Restrict caller identity support varies for proxies from different vendors m Use standard status messages otherwise messages will be taken from SIP packets Voice mail number or address Y Remove fancy characters from phone numbers O TO URL Port Fwd to External another device s P The Explanation of Picture e When your Mobile VolP are busying in line you can set up Fwd to External it s for you
9. number SIP Proxy Server or Asterisk needs to set the route of destination phone number 8 2 Route Mobile to LAN Speed Dial Settings When you set both Mobile to LAN Speed Dial Settings and Mobile to LAN settings at the same time Mobile to LAN Speed Dial Settings gets higher priority Mobile to Lan setting will be not available Mobile To LAN Speed Dial Route ME Name URL Select a 1 F F Mobile 3 E A E Network 5 SIP Settings 6 E NAT Transform T E Update 0 El 9 fai System Authority A Delete Selected Delete All Reset Reboot Add New Phone Position 0 9 Name URL ES Add Reset The call is answered with a prompt dial tone for the caller to press the Num and then the device connects the URL as destination Example after you call tte GSM number of the device and hear a dial tone you press 0 then the lan phone of IP address 192 168 0 107 rings 8 3 Route LAN to Mobile Settings In this page Lan To Mobile table you could set the routing rules to transfer the calls incoming from Lan to Mobile Maximum 50 sets TT i LAN To Mobile Table Page E noms Mobile To Lan Settings 0 3 3054396194 Mobile To Lan Speed Dial 1 y Mobile 3 m Network 4 F B SIP Settings z NAT Transform 7 a Update 3 r System Authority 9 E Save Change Reboot Delete Selected Delete All Reset Add New Position
10. number correctly After setting remember to click the Submit button TLL Ports Setting sone Mobile SIP Port 5060 1024 65535 Network RTP Port 60000 1024 65535 SIP Setti aia Submit Reset Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update _ System Authority Save Change Reboot 25 11 3 Codec Settings You can setup the Codec priority RTP packet length in this page You need to follow the ISP suggestion to setup these items After setting remember to click the Submit button Codec Settings LL WYTRON e Codec Priority Route Codec Priority 1 6 711 u law Mobile Codec Priority 2 6 711 a law Network Codec Priority 3 6 723 Codec Priority 4 6 729 SIP Settings ne Codec Priority 5 6 726 16 se Codec Priority 6 6 726 24 E Port Settings Codec Priority 7 6 726 32 Codec ID Setting Codec Priority 8 G 726 40 y DTMF Setting RPort Setting RTP Packet Length SIP Responses Other Settings G 711 amp G 729 20 ms NAT Transform G 723 30 ms y E System Authority G 723 5 3K Con Of Save Change Reboot Voice VAD Voice VAD COn Off Submit Reset 26 11 4 Codec ID Setting You can setup the Codec ID in this page LL Codec ID Setting You could set the value of Codec ID in this page Route Mobile Codec
11. to transfer the phone call to another designate device forward to next TB series device 17 9 4 Mobile SMS Agent SMS Agent Read received SMS YTRON Route Port Status Mobile Standby Rx List Mobile Status SMS Sender settings Dest Num Maximum Number of UCS2 chars for this text box is 70 A Message SIP Settings NAT T f ransrorm You have 70 UCS2 chars remaining for your description Update System Authority _ Send Now Save Change Reboot 1 Rx List Read received SMS 2 Dest Num the Receiver s phone number 3 Message Here is for you to key the message When you click Rx List you can view all received SMS as follows SMS Rx List Read Status ETT __Datelime 1 REC READ 006335114545 00 01 01 19 34 22 2 REC READ 096935306062 08 03 12 16 25 27 18 Click the serial no you can view message as follows SMS Reader CC MIT N I CI 886935386862 08 03 12 16 25 27 Mw Serial can send SMS and receive SMS Back Delete 10 Network In Network you could check the Network status configure the WLAN Settings LAN Settings and SNTP settings 10 1 Network Status Network Status information of current Network in this page 19 Route Mobile Network WAN Settings SNTP Settings SIP Settings Network Status Ethernet0 WAN Interface LAN Interface Type Fixed IP Client IP 192 169 113 Mask
12. 0 8 3 101 a rtpmap 0 pcmu 8000 a rtpmap 8 pcma 8000 a rtpmap 3 gsm 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 145 7331 branch z9hG4bK4C4315351FC84CA582D14FB8C25FC3BF received 192 168 66 145 rport 7331 From user_1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt tag as2a2fbf98 Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Type application sdp Content Length 242 v 0 53 o root 2737 2737 IN IP4 192 168 66 202 s session c IN IP4 192 168 66 202 t 0 0 m audio 13798 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 16 a silenceSupp off register issue The packet date from Asterisk as follows Please note user_1002 s display name don t appear So the website s Display Name is not available lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK590e92b551233a10a0ae71944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7 e45b773130f1fc945efcee502f84042 192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 10 REGIS
13. 192 168 66 145 7331 gt tag 677373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x Content Length 254 51 v 0 0 1001 4804366 4807851 IN IP4 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101 a rtpmap 0 pcmu 8000 a rtpmap 8 pcma 8000 a rtpmap 3 gsm 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv test 2 SoftPhone gt call 1002 Mobile VolP 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 29hG4bK4C4315351FC84CA582D14FB8C25F 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 5 Max Forwards 70 Content Type application sdp User Agent X Lite release 1105x Content Length 254 v 0 0 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP
14. 1fc945efcee502f84042 192 163 66 203 in 15000 ms asterisk1 CLI gt lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK672fa67f59c2223275f5ee236d27597a 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 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 z9hG4bK672fa67f59c2223275f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36dgf1 To lt sip 1002 192 168 66 202 gt 56 Call ID 7e45b773130f1fc945efcee502f84042 0M192 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 OPTIONS sip 1002 192 168 66 203 5060 SIP 2 0 Via SIP 2 0 UDP 192 168 66 202 5060 branch z
15. 254 Mii announces the current setting DNS Server in the Primary DNS field Default 192 168 0 1 Check Firmware 128 IVR announces the version of the sone A Set as DHCP 111 The system is changed to DHCP 10 Set Static IP 112xxx xx DHCP is disable and system is Address x xxx xxx changed to static IP type Enter IP address using numbers on the telephone keypad Use the star key when entering a decimal point 38 Set Network Mask 113xxx xx Must set Static IP first X XXX XXx Enter value using numbers on the telephone keypad Use the star key when entering a decimal point Set Gateway IP__ 114xxx xx Must set Static IP first Address X XXX XXX Enter IP address using numbers on the telephone keypad Use the star key when entering a decimal point Set Primary DNS 115xxx xx Must set Static IP first Server X XXX Xxx Enter IP address using numbers on the telephone keypad Use the star key when entering a decimal point 39 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 11 u Law G 11 a Law G 723 1 5 3k G 723 1 6 3k G 29A G 29A B 18 4 Voice Quality VAD CNG AEC LEC Packet loss 18 5 GSM TB 2001 Dual BAND 900 1800 MHZ Tri BAND 900 1800 1900 MHZ Quad BAND 900 1800 1900 850 MHZ 40
16. 8 0 100 the other settings e g SIP setting mac address remains and automatically restart UL Restore Default Settings You could click the restore button to restore the factory settings Route Mobile Restore default settings default Network SIP Settings _NAT Transform Update New Firmware Vececessccecsessceseecesssceessssesccesessesteteeses system Authority Save Change Reboot 36 16 Reboot In this page you could click the Reboot button to restart the system pr f Reboot System You could press the reboot button to restart the system Route Mobile Reboot system _ Reboot Network SIP Settings NAT Transform Update New Firmware Default Settings System Authority Reboot 27 17 Setting and checking via IVR User could get or set some parameters of the system by dialing in the mobile numbers of the device The status or result is reported via voice response system In the first 20 seconds after power on when only Mobile light flash you could dial its mobile numbers When you hear the dial tone press the following codes to set or check the device 195 Reboot the device Factory Reset 198 Return to default settings IVR announces the current IP address Default 192 168 0 100 Default off Check Gateway 124 IVR announces the current IP Address gateway IP address Default 192 168 0
17. 9hG4bK b92dd8a 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 5ebc2211278e2cb 69991 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 z9hG4bK672fa67f59c22232 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 57 Call ID 7e45b773130f1fc945efcee502f84042 0M192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Expires 300 Contact lt sip 1002 192 168 66 203 5060 gt expires 300 Date Tue 22 May 2007 03 11 54 GMT Content Length 0 58
18. ME TS CI EEES O Authorised Mobile 103 1 Another Authorised Mobile 103 C 2 3 4 5 6 7 5 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 _45 Service Domain Settings Realm 1 Default Active ON O OFF Display Name 103 User Name 103 Register Name 103 Register Password Asterisk Extension Password Domain Server Asterisk IP Proxy Server Outbound Proxy otatus Mot 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 Not Used Codec Priority 4 Not Used Codec Priority 5 Not Used wal Codec Priority 6 Not Used Codec Priority 7 Not Used Codec Priority 5 Not Used w 711465 723 RAE y RTP Packet Length 20 ms pi 30 ms G 723 5 3K LES MS i O On Off Voice VAD oice AD O On O Off 46 It Is very Important to use
19. TER 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 54 11 headers 0 lines Using latest REGISTER request as basis request Sending to 192 168 66 203 5060 NAT Transmitting NAT to 192 168 66 203 5060 SIP 2 0 100 Trying Via SIP 2 0 UDP 192 168 66 203 5060 branch 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 7e45b773130f1fc945efcee502 f84042 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 7e45b 773130f1fc945efcee502f84042 192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY WWW Authenticate Digest algorithm MD5 realm asterisk nonce 5def9231 Content Length 0 Ns Scheduling destruction of call Te45b773130f
20. alue or select the item according to your situation Note Please remember to save change whenever you submit any setting Click Save Change then Save button the system will restart and make the changed function setting operative 8 Route 8 1 Route Mobile to LAN Settings In this page Mobile To Lan Table you could set the routing rules to transfer the calls incoming from MOBILE to LAN Maximum 50 sets TIT Mobile To LAN Table Page 1 ERDRE E N LA l yy WA at ee Mobile To Lan Settings 0 z E Mobile To Lan Speed Dial 1 r Lan To Mobile Settings 2 IS Mobile 3 r Network 4 SIP Settings 6 E NAT Transform 7 r _ Update 3 r System Authority 9 r Save Change Reboot Delete Selected Delete All reset Add New Position 0 49 CID Ex 0911111111 0911 URL Ex 192 166 0 1 2St Add reset When the GSM number of the Mobile VoIP is called this device transfers the call to URL according to the caller ID of the incoming call 8 1 1 CID caller ID the numbers of incoming call You could set the CID as the following formats 1 The complete number e g 0911111111 2 The prefix part plus e g 0911 This format means any number starting with 0911 will be accepted to transfer 3 this means any incoming call is accepted to transfer 4 N this means the incoming call without showing its CID is accepted to transfer
21. date System Authority Save Change Reboot _29 11 7 SIP Setting SIP Responses LL SIP Responses Setting Route Response on port busy Mobile 486 Busy here Network C 503 Service unavailable A SIP Responses Service Domain ON OFF 180 Ringing Auto force to ON if 183 was OFF Port Settings CON OFF 183 Session Progress Codec Settings Codec ID Setting I DTMF Setting _Submit Other Settings NAT Transform Update System Authority Save Change Reboot 11 7 1 486 busy here 503 Service unavailable When Device are busying you can select 486 or 505 to response to SIP 11 7 2 180 Ring on off LAN TO MOBILE two stage dialing can be turn off therefore there will be no the Ring Back Tone all the phone call will be transferred to Voice Attempt directly For this function 183 must be turn on 11 7 3 183 Session Progress gt It means on progressing When you turn 183 on it means you can hear Voice Attempt while GMS side are busying We recommend you to turn this on if you use SIP Proxy 30 11 8 Other Settings You could setup the RFC and QoS according to your ISP information After setting remember to click the Submit button The QoS setting is to set the voice packets priority If you set the value higher than 0 then the voice packets get the higher priority to the Internet But the QoS function st
22. figure 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 66 145 331 username 1001 displayname user_1001 Mobile VoIP address 192 168 66 203 5060 username 1002 displayname user_ 1002 test pstn gt call 092849291 1 mobile number Mobile VoIP 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 50 To lt sip 1001 192 168 66 145 7331 gt Contact lt sip 1002 192 168 66 202 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 192 168 66 202 CSeq 102 INVITE User Agent Asterisk PBX Max Forwards 70 Date Tue 22 May 2007 02 50 37 GMT Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Content Type application sdp Content Length 242 v 0 o root 2737 2737 IN IP4 192 168 66 202 s session c IN IP4 192 168 66 202 t 0 0 m audio 15852 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 16 a silenceSupp off SIP 2 0 200 Ok Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001
23. ill need to cooperate with the others Internet devices TAL Other Settings Route Hold by RFC of Mobile C On Off Mobile Voice QoS 40 0 63 Network SIP QoS 40 0 63 SIP Settings SIP Expire Time 300 60 86400 sec Service Domain Port Settings Submit Reset Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses NAT Transform Update System Authority Save Change Reboot Aje 12 NAT Trans In this page NAT Trans STUN you could setup the STUN Enable Disable and STUN Server IP address This function helps your VolP device work properly behind NAT Change these settings according to your ISP information After setting remember to click the Submit button TAL STUN Setting Route STUN of Mobile C On Of Mobile STUN Server stunxtencom 2 0 Network STUN Port 3478 1024 65535 SIP Settings Submit Reset NAT Transform _Submit Reset STUN Setting system Authority _Save Change Reboot 13 System Auth In this page System Authority you could change your login name and password 39 System Authority You could change the login username password in this page New username New password fici A Confirmed password SIP Settings Submit Reset NAT Transform STUN Setting Update Save Change Reboot Route Mobile 93 14 Save Change P
24. 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 DIMF detection allow alaw 21 6 extensions conf GSM Gateway incoming calls gateway exten gt _103 1 Answer exten gt _103 2 DigitTimeout 3 give enough time to do second stage dialing exten gt _103 3 Response Timeout 5 exten gt _103 4 DISA no password outgoing here outgoing is the normal context to deal with the dial plan outgoing example of LAN to GSM call call the Mobile VolP sim card mail box thru GSM exten gt _888 1 SetCallerID XXXXXXXXXxX exten gt _888 2 Dial SIP EXTEN 103 60 r exten gt _888 3 Hangup 48 22 How to setup Asterisk to receive Caller ID from Test version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a X Lite 1105x 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 _49 qualify yes 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 con
25. lease remember this step whenever you submit any setting Click Save Change then Save button the system will restart and make the changed function setting operative TAL Save Changes i You have to save changes to effect them Route Mobile Save Changes Save Network SIP Settings NAT Transform STUN Setting Update System Authority Save Change 34 15 Update Here you could update the latest firmware and restore the default settings 15 1 Update New Firmware Update Firmware Download the latest firmware then 1 Method select HTTP 2 Code Type select Risc 3 File Location Click the Browse button in the right side of the File Location for the file Please note no need to unzip the firmware file 4 Click Update it takes few minutes to generate new firmware LL Update Firmware You could update the newest firmware PCB mark 2N149A Route Mobile Method HTTP TFTP Network SIP Settings HTTP ir Code Type Risc Update File Location Browse TE E Default Settings Uda TFTP Server 192 168 1 250 System Authority Save Change Update Reset Reboot Kaa pressi 35 15 2 Restore Default Settings In this page Update Default Settings you could restore the factory default settings to the system Click the Restore button then the system returns to default IP http 192 16
26. nction to this device Input the primary and secondary IP Address of SNTP Server to get the date time information Also you could set the Time Zone according to your location and set the time to synchronize After setting remember to click the Submit button LL SNTP Settings You could set the SNTP servers in this page Route Mobile SNTP On Cof Network Primary Server time windows com Status Secondary Server 208 184 49 9 SIP Settings Time Zone GMT E 08 oo hh mm NAT Transform Sync Time 1 i fo fo dd hh mm Update Submit Reset System Authority _Submit Reset Save Change Reboot DI 11 SIP Setting If you need you could setup the Service Domain Port Settings Codec Settings RTP setting RPort Setting and Other Settings If ISP provides the VolP service you need to input the related information correctly to register at SIP Proxy Server 11 1 SIP Setting Service Domain In this page you should input the data refer to your ISP Maximum is 3 accounts Realm 1 to 3 You could dial out via first SIP account and receive via the three SIP accounts LL Service Domain Settings Route Realm 1 Default Active ON OFF Mobile Display Name 201 a User Name 201 SIP Settings Register Name 201 Register Password Port Settings l Codec Settings Domain Server 192 168 1 1 Codec ID Setting Proxy Server
27. number with prefix of 09 When your lan phone and Mobile VolP had registered and 09 prefix is setted the routing rules at proxy server or Asterisk 4 d n a ppp this means to do the above routing and to modify the numbers Note d n means to delete the number of prefix a ppp means to add ppp prefix 11 E g d2a09 means to call the registered numbers via one stage dialing The numbers are modified to delete 2 digits of prefix of the original numbers then add 09 to be new prefix of the destination numbers 9 Mobile 9 1 Mobile Mobile Status In this page Mobile Status you could get the information of your GSM network and the latest operation TL Mobile Stato 2008 11 09 20 41 Route Operation Station AT amp T Mobile o SIM Card ID 89014104211834632187 Settings Signal Quality 17 Fwd Settings Registration State 0 1 SMS Agent GSM S N 352024020151233 Network SIP Settings Incoming URL IPX 192 168 1 1 NAT Transform Incoming Name IPX Update Outgoing IP 92888884 System Authority Incoming Mob 3058123530 Save Change Outgoing Mob 3054396194 Reboot 1 Network Registration The telecom carrier which the SIM card been registered 2 SIM Card ID SIM card ID 3 Signal Quality Signal quality 4 GSM S N IMEI Number 5 Incoming IP The IP address of the last incoming call from LAN 1 6 Incoming IP Name proxy server name 7 O
28. of Mobile VoIP 5 4 Plug the adapter in DC 12V socket and PWR socket The PWR light should turn red at the moment 5 5 Click reset button 3 sec Mobile VolP will restore default IP Other setting as usual 6 Setting and managing via web page The default IP address of Mobile VolP is http 192 168 0 100 Before accessing the web page please confirm this address is available in your network Login VoiP Enter your username and password to login VolP server Username Password Clear Enter the default username and password to login Default username voip Default password 1234 7 System Information 7 1 After login you could see the system information such as model name firmware version codec version name etc in this page LL Mobile VoIP x6 693 t WYTRON This page illustrate the system related information Route Mobile Model Type TB 2002 Model Description Transbox voip de 1 porta Network Firmware Version Mon Sep 8 08 59 42 2008 SIP Settings Codec Version Mon Jul 24 10 55 05 2006 Company Weytron CTI Corp _NAT Transform 2550 NW 72nd Ave Suite 108 Update Contact Address Miami FL 33122 Cod USA _System Authority Tel 305 436 7997 _Save Change _ Reboot 7 2 You could also see the setting table in the left side Please click on the option you would like to set The setting methods are indicated as the following chapters please input the v
29. 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 O 15 LAN Dialtone Gain 10 0 12 Mobile ON OFF Routing Range 0 to 49 0 49 CODEC Tx Gain 6 0 7 CODEC Rx Gain 6 0 7 SIP From Tel User Standard v Answer Delay 0 0 15 CLID Presentation O Suppression Invocation These settings seem to be ok just adjust 21 3 Antenna position Another important thing is to properly place the provided antenna If your gsm reception is good you should get around 18 or 19 as Signal Quality in the Mobile Status page With that level of signal quality your audio quality will be very good On the other end the signal quality down to 11 audio becomes very jerky So maximum signal quality maximum audio quality 21 4 Asterisk configuration Once the Mobile VoIP is set you have to configure Asterisk On that side you have to setup files as follow 21 5 sip conf GSM VOIP Gateway Mobile VoIP 103 _47 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
30. p 3 Set Route request Mobile to Lan 1 gt it is two stage dialing when mobile call in Mobile VoIP will provide dial tone and you can enter ip or asterisk extension or phone number If you want to enter phone number please note your asterisk need to have route of destination number 2 specific extension or IP or phone number when mobile call in Mobile VolP 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 Mobile VoIP will provide dial tone and you can enter mobile number 2 specific mobile number when lan phone call in Mobile VoIP will connect with the specific mobile number auto 3 gt It is 1 stage dialing When lan phone and Mobile VoIP 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 Mobile VoIP All changes both need to click save and change 42 21 Appendix Setup Mobile VoIP with Asterisk 21 1 Usage A typical usage of such a gateway is to be able to give a call with your normal mobile to any destination at voip cost Your mobile lt gsm network gt Mobile VoIP lt an gt Asterisk lt internet gt VOIP provider lt whate
31. response for setting and status enquiring Dial in GSM numbers of TB 2001 to get voice information or to operate 24 50 sets of LAN gt MOBILE routing and 50 sets of MOBILE gt LAN routing 2 5 Series connections to save bills 2 6 Standard SIP RFC2543 RFC3261 protocol to communicate with other gateways or PC 2 settings and managing via web page 3 The contents in package 3 1 Mobile VolP main body 3 2 AC DC Adaptor 110V AC 12V DC or 220V AC 12V DC 3 3 Network cable 3 4 Antenna 3 5 User s Manual When you receive TB 2001 package and find it is damaged or incorrect please contact your vendor 4 Dimension and Panel description 4 1 Antenna Antenna connector 4 2 DC 12V Power socket 4 3 LAN Standard RJ 45 socket connecting to Hub circuit 4 4 PWR Power indicator light red light Light is on when system s power supply is normal 4 5 MOBILE GSM indicator light green light Light flashes when GSM status is normal light turns on constantly when GSM is called 4 6 LAN LAN indicator light green light Light flashes when Lan is called light turns off when GSM answered 4 7 LINK Link indicator light green light Light is on when network is connected correctly 9 Accessory attachment 5 1 Connect the network cable both to your Hub and to LAN socket of Mobile VolP 5 2 Connect the antenna and place it in a good receiving location not too close to the device 5 3 Insert a SIM card into back
32. server please choose this option And how to transfer the caller ID to LAN please refer 22 How to setup Asterisk to receive Caller ID from Mobile VolP page 43 Mobile VolP will send the message as follows in the Packet From caller number lt sip 3001 192 168 0 228 gt tag 51088abb o Tel Tel Mobile VolP 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 and choose Active on else field empty in sip setting service demain O User Tel Mobile VolP will send the message as follows in the Packet From Username lt 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 and choose else field empty in sip setting service demain 8 Presentation CLIR If you need to block the Caller Id for call termination please choose Suppression 9 Mobile PIN Code If you need to unlock pin code via Mobile VoIP you can click On and enter pin code 15 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
33. utgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call from MOBILE 9Outgoing Mob The called number of the last outgoing call to MOBILE 13 9 2 Mobile Mobile Setting In this page Mobile setting you could adjust the parameter and click on the option to fit your need You could leave those default value before you had tried the complete operation of this device LL Mobile Setting Route 1 VolP Tx Gain a 0 12 2 VolP Rx Gain 111 0 15 Mobile 3 LAN Dialtone Gain b 0 12 4 ing 5 Routing Range fo to 49 0 49 SS 7 CODEC Tx Gain fe 0 7 6 CODEC Rx Gain 60m iti 8 SIP From Tel User Standard Answer Delay fo 0 15 12 SIP Settings 9 CLD Presentation Suppression Invocation NAT Transform 10 Mobile PIN Code On Tf Code Confirmed Update 1 LAN Answer Mode Answered Alerted Income system Authority Save Change Reboot Submit _Reset 1 VoIP Tx Gain To adjust the volume of LAN side 2 VolP Rx Gain To adjust the volume of Mobile side 3 LAN Dialtone Gain DIMF Reciver is not good you can adjust gain down 4 ON Off If you use this channel please click on Otherwise please click off 1 4 5 CODEC Tx Gain as above 6 CODEC Rx Gain as above 7 SIP From Caller ID transfer o Tel User Standard If you need to register to Asterisk and proxy
34. ver gt landline To do such a call you just call your Mobile VolP 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 Mobile VolP for free You can then call all around the world from your mobile at voip cost 21 2 Mobile VolP 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 Mobile VolP to work with Asterisk you need first to configure the box Here are some screen shots showing all the important parameters You have to note that in all the configuration process the Mobile VolP is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation _43 WAN Settings You could configure the YWAN settings in this page IP Type Fixed IP O DHCP Client O PPPoE IP Mobile VoIP IP Mask 255 255 255 0 Gateway Router IP DNS Server 1168 95 192 1 DNS Server2 168 95 1 1 MAC PPPoE Setting User Name Password LAN To Mobile Table Page 1 Y e IWRIkf O Your Asterisk IP 1 2 al A 5 6 E 5 al Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM 44 Mobile To LAN Table Page 1 Y

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