User Manual Model: GoIP1
Contents
1. will be added in front of the number when the first digit of the dialed number is 1 a Number entered 1082121234 actual number dialed 00182121234 b Number entered 82121234 actual number dialed 82121234 3 Dial Plan 001 001 1751 means that the first 3 digits 001 of the dialed number will be changed to 1751 when a number with first three digits 001 is entered a Number entered 00182121234 actual number dialed 175282121234 b Number entered 82121234 actual number dialed 82121234 4 Dial Plan XXXX means that the input number is limited to 4 digit long and will be dialed out immediately when the fourth digit is entered 5 Dial Plan 13XXXXXXXXX 0 means that the input number is restricted to 11 digit long and the first two digits must be 13 When this condition is matched the digit 0 will be added to the front of the number and then dialed out a Number entered 13901234567 actual number dialed 013901234567 b Number entered 12801234567 actual number dialed 12801234567 6 Dial Plan 13 6 9 XXXXXXXX 0 means that the input number is restricted to 11 digit long the first two digits must be 13 and the third digit can be 6 7 8 or 9 When this condition is matched the digit 0 will be added to the front of the number and then dialed out a Number entered2. VAD Innovative Sending Message to 192 168 2 1 5060 INVITE sip 9998675588228822 192 168 2 1 5060 transportudp SIP 2 0 Via SIP 2 0 UDP 192 168 2 189 5060 rport branch z9hG4bK1686911003 From lt sip 8613800000000 192 168 2 1 5060 gt user phone tag 626918067 To lt sip 9998675588228822 0192 168 2 1 gt lt Call ID 1835068843 0192 168 2 189 5060 CSeq 2 INVITE Contact lt sip 6138000000000192 168 2 189 5060 gt Max Forwards 30 User Agent HyberTone Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 One Channel GSM VoIP Gatewa Mode 2 example GoIP1 use SMS Dial Mode 2 PSTN Trust List gt GSM Group Mode SMS Mode SMS Dial Mode 2 S sms piatPrefs A mobile phone s number is 86 13800000000 it sends a SMS 8675588228822 to GolP1 s GSM SIM card When GolP1 device receives this SMS it will automatically call the number 8675588228822 and the caller number is GolP1 s SIP account number GolP1 will set the SMS sender number to Call Divert option s Forward Number VolP to PSTN automatically The result is when SIP server receives the SMS call and call back to GolP1 GolP1 will automatically call the SMS sender via GSM network The sent out signaling as follow 32 VAD Innovative Sending Message to 192 168 2 1 5060 INVITE sip 8675588228822 0192 168 2 1 5060 transportudp
3. 13901234567 actual number dialed 013971234567 b Number entered 13001234567 actual number dialed 13001234567 Please note that the above samples are intended to show the meaning of various rules They may not have any practical meaning A combination of these rules joined with the symbol can be realized for a much more complicated dialing application 3 9 Gain Settings A hidden webpage is provided to set the receiving and transmitting gains of VolP Chunnel The URL link is http Xxx xxx xxx xxx vadcore en_US gain html THIS PAGE IS INTENDED FOR AN EXPERIENCED USER OR AN ADMINISTRATOR ONLY PLEASE 39 Vanes One Channel GSM VoIP Gatewa SET THE GAINS WITH CAUTIONS Note A too low or too high gain MAY affect the sensitivity of DTMF detections Line 1 Line 1 Output Gain 3 10 Network Configuration Line 7 Input Gain 2 oS os Click on Network tab in the left menu column to configure the LAN and PC ports Network Configuration LAN Port Static IF s IP Address 192 168 0 100 Subnet Mask optional 255 255 255 0 Default Route 192 168 0 1 Primary DNS EH optional 802 1q VLAN Enable Disable Advanced gt 3 10 1 LAN Port IP Address Subnet Mask DHCP Server Enable Disable Starting Address 192 168 8 100 Ending Address 192 168 585 120 Static DNS optional sd Advanced gt gt Three LAN Port modes are supported DHCP
4. Channel GSM VolP Gatewa Connect to 192 168 0 100 The server 192 168 0 100 at Please Login requires a Username and password warning This server is requesting that your username and password be sent in an insecure manner basic authentication without a secure connection Remernber my password 3 2 Status The Status page shown below is the default home page of the GoIP1 Web server Innovative Status Status Phone Information Network Information GSM Module Information Configurations Serial _ GOlP09100621 LAN Pen 192 168 0 100 GSM Model SIMCOM SIM300 SE deiere GHS 3 01 36 LAN MAC 00 11 BE 02 F5 12 GSM SIM NOT INSERTED 1ools mer eee GSM dwar PC Port 192 168 8 1 ee Got ji Operator weem PPPoE Disabled Phone Status LOGOUT Gen SEH oe ae Route 192 168 0 1 GSM Status LOGOUT DNS Server GSM Band 850 1900 SIM Remain Time NO LIMIT 11 VAD Innovative One Channel GSM VoIP Gatewa 3 2 1 Phone Information Serial Number Each Gateway has a unique serial number assigned by the factory such as GOIP109100019 This number is important for centralized configuration technical support and warranty This number is printed on the bottom of the Gateway and is associated with your software license Firmware Version Firmware version identifies the firmware version of the Gateway such as GHS 3 01 36 Hardware Mode This field shows terminal s hardware type Phone Status This field sho
5. SIP 2 0 Via SIP 2 0 UDP 192 168 2 189 5060 rport branch z9hG4bkK92531 725 From lt sip 20001 192 168 2 1 5060 gt user phone tag 740569827 To lt sip 8675588228822 d192 168 2 1 gt lt Call ID 464713443 0192 168 2 189 5060 Cseq 3 INVITE Contact lt sip 20001 4192 168 2 189 5060 gt Max Forwards 30 User Agent HyberTone Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 One Channel GSM VoIP Gatewa SMS prefix can be used in mode 2 just like in mode 1 Mode 3 example GoIP1 use SMS Dial Mode 3 PSTN Trust List gt gt GSM Group Mode SMS Dial sms Dial Prefix A mobile phone s number is 86 13800000000 it sends a SMS 8675588228822 to GolP1 s GSM SIM card When GolP1 device receives this SMS it will automatically call the number 8675588228822 86 1 3800000000 and the caller number is GoIP1 s SIP account number The sent out signaling as follow 33 VAD Innovative Sending Message to 192 168 2 1 5060 INVITE sip 8675588228822 S6 1 38000000000192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 180 5060 branch z9hG4bK620642232 From lt sip 20001192 168 2 1 5060 gt user phone stag 19399994780 To lt sip 8675588228822 86138000000000192 168 2 1 gt Call ID 52754291 192 168 2 180 Cseq 2 INVITE Contact lt ip 20001 192 168 2 180 5060 gt Max F orwards 3
6. add and the port number For example sip hybertone com 8080 E Register Expiry s F G Enter the register time seconds in this field This is the maximum length of registration that SIP server will keep your registration If SIP server does not receive another SIP registration the current registration will time out Check your SIP server for a reasonable value Outbound Proxy Outbound proxy is a device that receives requests from a client even though it may not be the server resolved by the Request URI Outbound proxy will forward SIP requests and frequently RTP media traffic to another SIP server Outbound proxy is used for a number of reasons including firewall traversal both in parallel with a firewall and situated in the Internet as a Session Border Controller and also for hiding customer IP addresses calls are all routed through one point so that a public ITSP IP address can be used for accessing customers rather than the customer s own IP address Check with your SIP server SIP provider if an outbound proxy is required Home Domain SIP Networks sometimes use the Home Domain name as an identifier Enter this field if it is required H Authentication ID l Enter the Authentication ID as provided Password Enter the authentication password as provided J Call Forward Type K L Call forward can be set under different conditions Unconditional Forward Busy Forward No Answer Forward Busy
7. be used and its priority for a voice call Audio Codec Preference alaw ulaw g729 g729a g729ab g7231 25 V D Innovative One Channel GSM VoIP Gatewa Click on the check box to enable a codec Select a codec and then press the UP or DOWN button to move the position of the codec on the codec list with a priority in descending order Note The voice code alaw and ulaw is G 711a and G 711u 3 5 Call Divert The Call divert feature controls the routing of calls between VoIP and GSM Call Divert Forward to PSTN Enable Disable SMS Mode Disable v at oppenen Mode Use CID as SIP Caler v VoIP To PSTN CID Forward Mode Use CID as SIP Caller See TF PSTN ete N Noah e SIMCardExpiry o oZ o y Report Number SIM Card State Report Time smcaao OOS VoIP Trust List gt gt Forward to VolP Enable Disable e a PSTN To VolP EH VoIP Forward to VoIP Auth Mode No Auth w PSTN Trust List gt 3 5 1 Call Forward From VoIP To PSTN 26 V D Innovative One Channel GSM VoIP Gatewa Call Divert Forward to PSTN Enable Disable eee EH VoIP To PSTN a Ce PSTN deieren Auth Mode a K SIM Card Expiry i O enee Report Number a Report Time smcardio Sd VoIP Trust List gt Forward Number Enter a phone number in this field will forward all incoming VoIP calls to this phone number PSTN or Mobile Using to add a 500ms delay to the dialing seq
8. contact VADcore for more details 3 4 Call Settings Click on the Call Settings to configure the VoIP call settings VAD Innovative 3 4 1 SIP Standard Supported One Channel GSM VoIP Gatewa GoIP1 supports SIP standard GolP1 has two types of config modes for SIP protocol Call Settings Config Mode SIP Trunk Gateway1 amp SIP TrunkGateway2 i Single Server Mode The channel uses a SIP account to connect to SIP server Trunk Gateway Mode The GolP1 will act as a SIP proxy Remote SIP clients can register to GoIP1 and GoIP1 will process SIP requests on behalf of SIP client GoIP1 Gateway s SIP configure page as follow Call Settings Config Mode Single Server Mode Phone Number fe Display Name Po SIP Proxy SIP Registrar Server Ps Register Expiry s Outbound Proxy Po Home Domain Po Authentication ID SY Password Call Forward Type Call Forward Number Oooo O Backup Server O Enable Disable A Phone Number Enter a SIP phone number B Display Name Enter this field for the name to be displayed on the called VolP party C SIP Proxy VAD Innovative One Channel GSM VoIP Gatewa Enter the SIP proxy IP address or domain name If the registration port is not 5060 then add and the port number For example sip hybertone com 8080 D SIP Registrar Server Enter the SIP registrar server IP address or domain name in this field If the registration port isn t 5060
9. for Signaling NAT Traversal Enter the IP Address or the domain name of the STUN server to be used 3 Port forwarding Support Port forwarding sometimes referred to as tunneling is the act of forwarding a network port from one network node to another This technique can allow an external user to reach a port on a private IP address inside a LAN from the outside via a NAT enabled router In order for this feature to work the local network Gateway must support this feature and be set up properly Please consult your network administrator for help to enable this Port forwarding feature 24 VAD Innovative 4 Relay Proxy Relay proxy is a proprietary NAT traversal technology Please consult your service provider for more information Currently the following 3 kinds of packaging mechanism are supported gt Mode 1 The media uses UDP packets and or encrypt with multiple UDP port gt Mode 2 The media uses UDP packets and or encrypt with single UDP port gt Mode 3 The media uses TCP packets and or encrypt UDP over TCP One Channel GSM VoIP Gatewa Media NAT Traversal User Name Po Password d Encryption Relay Mode Backup Relay Server 1 Backup Relay Server 2 Backup Relay Server 3 Backup Relay Server 4 3 4 5 Codec Preference Click on Media Settings in the Call Setting menu and click Audio Codec Preference to access the parameters Codec Preference allows a user to select the codes to
10. if you need other language support 3 3 2 Time Zone and Time Server The GolP1 Gateway supports Network Time Protocol NTP to obtain the date and time information from an NTP server Time Server The time zone is specified as in GMT offset For example the Pacific Standard Time is GMT 8 and the Pacific Daylight Time is GMT 7 Time fone GMT 8 Time Server pool ntp org Note The GolP1 Gateway supports CDR and Billing Information it is important to set up these two parameters properly 3 3 3 DTMF Min Detect Time Gap DTMF Min Detect naaf Time Gap This parameter is used to limit two same DTMF digit s minimum time gap the range is 60ms to 120ms default is 80ms If you encounter double digit problem increase this parameter If you encounter lose digit then decrease this parameter 3 3 4 Network Tone Network Tones are a set of tones used for VolP calls Select one of the predefined countries 14 V D Innovative One Channel GSM VoIP Gatewa or select Customized to define your own Network Tones Network Tones China Australia China Hong Kong New Zealand United Kingdom United States Customized You can configure the Network Tones as Customized option Network Tones Dial Tone fe Ring Back Tone fs Indication Tone ee Each tone listed above is defined in the following format nc rpt c1on cloff c2on c2off con c3off f1 f2 f3 f4 p1 p2 p3 p4 Where nc is the number of cad
11. network equipment c Optional Connect an Ethernet cable to the PC Port of the GoIP1 Gateway and the other end to a PC or other network device d Connect the power adapter provided to the power jack of the GoIP1 Gateway VAD Innovative 2 2 LED Indicators One Channel GSM VoIP Gatewa The following table defines the status of the LEDs located on the top case and on the RJ 45 connectors When the GolP1 is booting this LED will flash 100ms ON and 100ms OFF When the GolP1 is properly registered to your softswitch this LED flashes at a rate of 1s ON and 1s OFF GSM When the GSM channel is ready to sue this LED flashes at a rate of 1s ON and 1s OFF 2 3 SMS Commands GolP1 supports some maintenance commands from SMS FUNCTION SMS CONTENT REMARK Note In command Reset and Reboot the Password is the GoIP1 device s admin password The command keywords can be uppercase and lowercase but the password is case sensitive 1 Obtain LAN Port IP Address Once the GSM SMS with message content info or INFO is received the GoIP1 sends back a SMS message to the sender with the message content containing the LAN Port IP address 2 Reset GoIP Configuration Upon receiving the SMS message RESET lt password gt or reset lt password gt the GolP1 resets its configurations to factory defaults 3 Reboot GoIP Upon receiving the SMS message REBOOT lt password gt or reboot lt password g
12. or No Answer Forward Select the call forward type and enter the phone number that you would like the call to be forwarded to Call Forward Number Enter the phone number that you would like the call to be forwarded to when Call Forward is set Backup Server VV A B Innovative One Channel GSM VoIP Gatewa Backup Server Enable Disable eler Registrar oe Domain Fail retry Interval 1 IT E 60s The GolP1 Gateway supports one Backup Server as an alternative to the main server When the registration to the main server fails the GolP1 Gateway will try to register to the Backup Server 3 4 2 Advanced Settings Click on Advance Settings tab on the top right corner of the Call Setting page to display all the parameters for programming as shown below These parameters allow more advanced control over the SIP signaling and media preference Advanced Settings lt lt Local Signaling Port CI SIP 183 NAT Keep alive Enable Disable Advanced Timing Outband DTMF type RFC 2833 RTP Payload Type 101 w sda Traversal Media Settings gt gt A Local Signaling Port SIP Local port The default SIP port is 5060 Change this as required B SIP 183 Check the box of SIP 183 if the SIP server supports this feature C NAT Keep alive 20 VAD Innovative NAT Keep alive Enable Disable One Channel GSM VoIP Gatewa The NAT Keep alive feature sends a null packet to the SIP Pr
13. server for Signaling NAT Traversal Enter the IP Address or the domain name of the STUN server to be used 2 Relay Proxy 22 V D Innovative One Channel GSM VoIP Gatewa Relay proxy is a proprietary NAT traversal technology Please consult your service provider for more information 3 4 4 Media Setting Click on Media Settings in the Call Setting menu to access the parameters available for media settings Media Settings lt lt RTPPortRange ez Es PacketLength ms 20 lt Jitter Buffer Fixed Delay ms 60 Media QoS None I Media Encryption None d symmetric RTP Media NAT Traversal None Audio Codec Preference gt gt A RTP Port Range This parameter specifies the range of the RTP Real Time Protocol Ports used by the GoIP1 Gateway If your network limits the usable port range this parameter may need to be modified Please consult your network administrator for more information B Packet Length ms This parameter defines the voice packet length The default setting is 20ms The range is from 5ms to 40ms at an increment of 5ms Please note that some codes have a minimum packet length of more than 5ms C Jitter Buffer Mode Jitter Buffer Mode Fixed 3 limit Since data packets may arrive at different orders the Jitter Buffer is used to hold the data packets received for re arrangement according to the packet sequence number Three Jitter Buffer Modes are supported Adaptive Se
14. 0 User Agent HyberTone Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 One Channel GSM VoIP Gatewa SMS prefix can be used in mode 3 just like in mode 1 3 6 2 SMS Relay GoIP1 GSM Gateway supports SMS relay PSTN Trust List gt gt GSM Group Mode Disable wi SMS Mode a sd Number The SMS Forward Number is the receiver ex an extension on your VoIP system GoIP1 will forward the SMS it received to the number 3 6 2 1 SMS Relay To VoIP System When GolP1 receives a SMS from GSM network it will relay to VoIP system s appointed number SMS Forward Number Assume the SMS Forward Number is 3999 and SMS sender number is 8613682626865 the SMS content is 075583185700 The GoIP1 will send a message to your VoIP system as below 34 VAD Innovative MESSAGE sip 3999 192 168 2 1 SIP 2 0 Via SIP 2 0 UDP 192 168 2 162 5060 branch z9hG4bK196 7685528 From lt sip 1638 192 168 2 1 gt tag 66 7435795 To lt sip 3999 192 168 2 1 gt Call ID 2094144847 192 168 2 162 CSeq 4 MESSAGE Contact lt sip 1638 192 168 2 162 5060 gt Max Forwards 30 User Agent HyberTone Content Type text plaine Content Length 28u 8613682626865 075583185700 One Channel GSM VolP Gatewa 3 6 2 2 SMS Relay To GSM Network When GolP1 receives a message from SIP server as below MESSAGE sip 1638 192
15. 168 2 162 5060 SIPF 2 0 From lt sip 3999 192 168 2 89 gt tag 5031 To lt sip 1638 192 168 2 1 gt Call ID 808807EB A8B3 DD11 BBA6 005056C00008 192 168 2 89 CSeq 3 MESSAGE Contact lt sip 3999 192 168 2 89 gt max forwards 16 date Tue 18 Nov 2008 06 36 37 GMT user agent SIPPER for 3CX Phone p hint usrloc applied Content Type text plain Content Length 26 13682626800 Hello world The GolP1 will send a SMS to GSM number 13682626800 the SMS content is Hello world 35 VAD Innovative 3 7 GSM Caller ID Transparent One Channel GSM VoIP Gatewa GoIP1 supports GSM Caller ID transparent to VoIP via SIP Invite signaling CID Forward Mode Use CID as SIP Caller Disable A Disable Disable GSM Caller ID transparent to VoIP B Use Remote Party ID GoIP1 will add Caller ID in SIP invite s Remote Party ID option Sending Message to 192 168 2 1 5060 INVITE sip 5000 192 168 2 1 5060 transportudp SIP D Via SIP 2 0 UDP 192 168 2 180 5060 branch z9hG4bK1645487913 From lt sip 20001 192 168 2 1 5060 gt user phone tag 406202416 To lt sip 5000 192 168 2 1 gt Call ID 847230278 192 168 2 180 CSeq 2 INVITE Contact lt sip 2000 192 168 2 180 5060 gt Max F orwards 30 User Agent HBT Remote Party ID 13800000000 lt sip 13800000000 192 168 2 1 gt party calling screen no privacy off Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INF
16. Change Password Reset Config Reboot 3 13 1 Online Upgrade To perform a firmware upgrade select the Online Upgrade tab to access the page below 44 V A B Innovative One Channel GSM VoIP Gatewa Online Upgrade Last Upgrade Time Current Version GH35 3 01 36 Upgrade Site Il T sg Enter the update link as provided by VADcore A sample link is hk ippcn com update GHS 3 01 18 pkg Click the Start button to start the firmware upgrade WARNING POWER SHUTDOWN POWER FAILURE OR UNPLUG POWER ADAPTOR FROM GolP1 DURING FIRMWARE UPGRADE MAY PERMINENTLY DAMAGE THE GOIP1 GATEWAY AND VOID THE WARRANTY 3 13 2 Change Password Click on the Change Password tab to access the page below User Level NewPassword 1 Confirm Password 1 Change _ Administration Level NewPassword Confirm Password 1 Lia A User Password This is the password for the user ID user The default password is 1234 This user ID has limited access to the Network Configuration menu B Administrator Password default admin This is the password for the user ID admin The default password is admin This user ID has full access to all configuration settings available 45 VAD Innovative One Channel GSM VoIP Gatewa 3 13 3 Reset Configuration Click on the Reset Config tab to reset the GoIP1 Gateway to its factory default settings 3 13 4 Reboot the Device Click on the Reboot tab to reboot the GoIP1
17. Control gt gt Network Configuration LAN Port Static IP v i IP Address 192 168 0100 Subnet Mask optional 255 255 255 0 Default Route 192 168 0 1 PrimaryDNS Ss tee e S optional 802 1q VLAN O Enable Disable Advanced gt gt Call Settings Config Mode Trunk Gateway Mode SIP Trunk Gateway Sd SIP Trunk Gateway2 sd SIP Trunk Gateways O Ei E Phone Number Register Expiry s Authentication ID il Password Network Tones United States v GSM Group Mode Disable v GSM CalleriD Anonymous O Enable Disable GSM Band 850 1900 EN SMS Sender O Enable Disable PC Port Static IP v IP Address Subnet Mask DHCP Server Enable Disable Starting Address Ending Address StaticDNS optional sd Advanced gt gt Advanced Settings gt gt Media Settings gt gt Click on Preference in the left menu of the configuration web and the screen will be displayed as below 13 VAD Innovative One Channel GSM VoIP Gatewa Preference Time Zone Network Tones United States Time Server pool_ntp org GSM Group Mode Disable J DTMF Min Detect GSM CallerlD e WEE Time Gap Anonymous J Enable amp Disable Remote Contral gt gt GSM Band 850 1900 k SMS Sender Enable Disable 3 3 1 Language Currently GolP1 only supports English VADcore also has other versions of software that support Simplified Chinese and Traditional Chinese Contact VADcore
18. DSLs Choose this if your network environment requires Enter the User Name and Password as provided by your ISP 4 802 1q VLAN This QoS feature requires QoS support of your network to improve voice traffic Please consult your network administrator for proper settings 5 Advanced The Advanced settings allow the user to set the broadcast address and to clone a MAC address instead of using the factory preset MAC address Please consult your network administrator for further information 3 10 2 PC Port Configurations The PC Port allows other network devices to be attached to the GolP1 Gateway It offers both Bridge and Static IP modes to meet your requirements The factory preset is Static IP mode with the IP address 192 168 8 1 42 V D Innovative One Channel GSM VoIP Gatewa PC Port Static IP kl IP Address 192 168 8 1 Subnet Mask 255 255 255 0 DHCP Server Enable Disable Starting Address 192 168 8 100 Ending Address Static DNS optional sd Advanced lt lt a O Address IP Broadcast Address 1 Bridge Mode Select Bridge mode if your network topology requires the network devices PC or others to be in the same network segment as the GoIP1 Gateway In this case the GolP1 Gateway functions as an Ethernet Switch 2 Static IP Mode Default Setting Select Static IP mode for a new network segment on PC port In this case the GolP1 Gateway functions as Router Enter the IP address in IP Address
19. Gateway 46 VAD Innovative One Channel GSM VoIP Gatewa 4 Hardware Specifications Processor Processor ARE 133MHZ 133MHz DC12V 500mA 10 Input AC100V to AC240V Default 850MHz 1900MHz aia tae RE DEEN e E a mr Igel 47 VAD Innovative One Channel GSM VoIP Gatewa 5 Manufactory Parameters 192 168 0 100 Network PC Static IP 192 168 8 1 DHCP Server Running Password Time Zone GMT 8 48
20. O SUBSCRIBE Content Type application sdp Content Length 226 C Use CID as SIP Caller ID GolP1 use PSTN Caller ID in SIP invitee s Caller ID option and Remote Party ID option 36 VAD Innovative Sending Message to 192 168 2 1 5060 INVITE sip 5000 192 168 2 1 5060 transportudp SIP 2 0 Via SIP 2 0 UDP 192 168 2 180 5060 branch z9hG4bk1450498491 From 13800000000 lt sip 13800000000 192 168 2 1 5060 gt tag 232569343 To lt sip 5000 192 168 2 1 gt Call ID 18530689860 192 168 2 180 CSeq 2 INVITE Contact lt sip 13800000000 192 168 2 180 5060 gt Max Forwards 20 User Agent HBT Remote Party ID 13800000000 lt sip 138000000000192 168 2 1 gt party calling screen no privacy offe Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type applications dp Content Length 226 One Channel GSM VolP Gatewa 3 8 Dial Plan Dial Plan defines how a number is processed when GolP1 receives it This field is located in the Call Divert Window The Dial Plan is very flexible and can be configured for a wide range of dialing applications 37 V D Innovative One Channel GSM VoIP Gatewa Call Divert Forward to PSTN Enable Disable ae KH Auth Mode No Auth s SIM Card Expiry o o i SIM Card State Report Number E SIM Card State SS Report Time B30 0000 VoIP Trust List gt Forward to VoIP Enable Disabl
21. Static IP and PPPoE The default is set for Static IP with default IP address 192 168 0 100 40 V D Innovative Network Configuration LAN Pert One Channel GSM VoIP Gatewa DHCP IP Address Static IP Subnet Mask PPPoE optional a Default Route 192 168 0 1 Primary DNS lL optional 802 1q VLAN Enable Disable Advanced ll Address IP Broadcast Address 1 DHCP Choose DHCP if a local DHCP host is available This allows the GolP1 Gateway to obtain network information IP Address Subnet Mask Default Route Primary DNS Secondary DNS and other DHCP options from the DHCP host 2 Static IP Network Configuration LAN Port Static IF Si 192 168 0 IP Address 9 168 0 100 subnet Mask optional 200 200 200 0 Default Route 192 168 0 1 W Po Secondary DNS optional E eegen ia Enable Disable Advanced lt lt Ethernet MAC Address IPBroadcastAddress sd The default setting of GoIP1 is Static IP with IP address 192 168 0 100 and Subnet Mask 255 255 255 0 However Default Route Primary DNS and Secondary DNS optional must be manually entered according to your network configuration 3 PPPoE 41 VAD Innovative One Channel GSM VoIP Gatewa Network Configuration LAN Por User Name Po Password ee 802 1q VLAN Enable Disable Advanced Fe Oo t izC Address PPPoE is a common method for you network modem Cable x
22. User Manual One Channel GSM VoIP Gateway Model GolP 1 Version 1 0 2010 03 Tel 1 650 963 7600 sales vadcore com Innovative support vadcore com VADcore REES www vadcore com VAD Innovative One Channel GSM VoIP Gatewa Contents 1 Product IDET ee el e e sasaesetececesecessssencastuncacsceovusessesasseesseaececeanssseensianzencpasccensseoeesssasene 3 t1 General ue et MAC OM EE 3 D r POCO e veigeccccneccceesscnceneccceesccues os5 ces oeeusene EAE T 3 1 3 Hardware Specification WEE 4 1 4 Software Specification ssssesseeeccesossseeeeeeceeccccossseceeeceeceecocccoseeeeceeceeccsoseseeeceeceeoessssee 4 1 3 List of the Package eege 4 AGO PDD Tue EE 5 EE 7 2 1 IEN e e nee CN 7 2 e NED NG CACO EE 8 Tod SIMS COMMING E 8 3 Configuration EE 9 3 1 Web Configuration MONU cccccccccccssssscsscssccccsscssscscssssscsscssssscsscsscescesescceccecscescesoes 10 E d 11 3 2 1 Phone Miornnat NEE 12 3 2 2 Network uge un C e sepson ene innne anre EE AENEAS E ENE ANAE AREE TETEE E 12 3 2 3 GSM Module Information 12 EEGEN 13 E e WEE 14 3 3 2 Time Zone and Time Server 14 3 3 3 DTMF Min Detect Time Gap EEN 14 3 34 Network Ee 14 BV Ee Le tel 16 3 3 6 GSM Caller ID ANONYMOUS 0 0 0 0 ceeccccccessscecesesseeecceesseeecceessssecesesseeceessseeceesssseeceessssesesenaes 17 KZ N oc E 17 EE Eeer 17 3 4 Gall EE 17 3 4 1 SIP Standard SUD DONE CG sais vesisiunisesnsivunsesiozisicvioustaleeswicestesieldhisesnainssal
23. e Forward Number PSTN To VoIP Dial Plan PSTN to VoIP Forward to VoIP Auth Mode PSTN Trust List gt The basic syntax is lt event gt lt action gt lt event gt lt action gt where lt event gt defines the event to be matched An event consists of a sequence of digits Ifa specific digit has a limited range use the syntax A B where A and B are both digit 0 to 9 and B is greater than A The length of the input number can be limited by using X to represent each unknown digit If this field is omitted it means any event lt action gt defines the action to be taken when a phone number is received It consists of minus plus and digits followed by digits means to remove the digits from the beginning of the number followed by digits means to add the digits in front of the number means or and the order of priority is from left to right Note For practical use there should be no limitation on the length of dial plan string Examples 1 Dial Plan 010 010 means that the first 3 digits 010 of dialed number will be removed if the first 3 digits of dialed number are 010 a Number entered 01082121234 actual number dialed 82121234 38 V D Innovative One Channel GSM VoIP Gatewa b Number entered 82121234 actual number dialed 82121234 2 Dial Plan 1 00 means that two digits 00
24. e in order for this to work properly GolP1 Gateway supports both RFC2833 and SIP INFO DTMF protocols 2 DTMF Payload Type DTMF Payload Type is defined by RFC2833 protocol to carry the tone definitions for various applications The default DTMF Payload Type is 101 Please consult your VolP service provider for the proper setting if required C Signaling Qos Signaling QoS 2 None N IP TOS DiffServ Signaling QoS improves the performance of SIP signaling If local network device supports Qos select this field accordingly Please consult your network administrator for further information D Signaling Encryption E GoIP1 Gateway supports different encryptions for SIP signaling Select the one that you prefer Signaling NAT Traversal Signaling NAT traversal may be required if the GolP1 Gateway is put behind a NAT firewall or multiple NATs firewalls Depending on your network environment and SIP Server capabilities this feature may or may not be turned on Signaling NAT Traversal STUN RFC 3489 Relay Proxy 1 None Select None to turn off this feature 2 STUN RFC 3489 STUN Simple Traversal of UDP User Datagram Protocol through NATs Network Address Translators is a network protocol allowing a client behind a NAT or multiple NATs to find out its public address the type of NAT it is behind and the internet side port associated by the NAT with a particular local port Select STUN RFC 3489 to use a STUN
25. e of VoIP is greatly enhanced with significant savings on long distance and roaming charges SS BS Local PSTN 5 mg Ei GSM Network Network IP Phone j i Network N a N D L Telephone Cellphone Telephone 1 2 Protocol TCP IP V4 IP V6 auto adapt ITU T H 323 V4 Standard H 2250 V4 Standard H 245 V7 Standard H 235 Standard MD5 HMAC SHA1 ITU T G 711 alaw ulaw G 729A G 729AB and G 723 1 Voice Codec RFC1889 Real Time Data Transmission Proprietary Firewall Pass Through Technology SIP V2 0 Standard Simple Traversal of UDP over NAT STUN Web base Management PPP over Ethernet PPPoE PPP Authentication Protocol PAP VV WWW WWW WW WW WV VAD Innovative Internet Control Message Protocol ICMP TFTP Client Hyper Text Transfer Protocol HTTP Dynamic Host Configuration Protocol DHCP Domain Name System DNS User account authentication using MD5 Out band DTMF Relay RFC 2833 and SIP Info One Channel GSM VoIP Gatewa VV VV VV WV 1 3 Hardware Specification ARMGQE Processor DSP for voice codec and voice processing Two 10 100 BaseT Ethernet ports with full compliant with IEEE 802 3 LEDs for Ethernet port status One GSM Channels Connection VV VV WV 1 4 Software Specification LINUX OS Built in HTTP Web Server PPPoE Dial up NAT Broadband Router Functions DHCP Client DHCP Server Firmware On line upgrade PSTN Caller ID transmit Multiple Language Support Support
26. ed call divert Supported PSTN auto call out to PSTN Supported Multi_devices Cooperate Mode Group Mode Supported SMS call out VV VV VV VV VV VV WV 1 5 List of the Package a One GolP1 Gateway main unit b One DC12V 500mA power adaptor c One Ethernet cable 3M VEH 1 6 Appearance One Channel GSM VoIP Gatewa VoIP GSM Gateway GoIP1 Front View VoIP GSM Gateway GoIP1 Rear View VAD Innovative One Channel GSM VoIP Gatewa 1 2 3 4 LAN Connect this port to an Ethernet Switch Router the Ethernet of a DSL modem or other network access equipment PC Connect a computer or other network device to this port POWER DC12V 500mA Connect the 12V 500mA Adapter provided to this power jack Reset Press this button to reset the GoIP1 Gateway to factory defaults V Mm DW Innovative 2 Installation One Channel GSM VoIP Gatewa 2 1 Installation Steps menage i CH Ze GSM Network 1 Pi s W Dem External Network Connection AC DC Power xDSL Cable Local Network Adapter Use the Modem Router Connection PC ie idad provided or Gateway etc Other Network equivalent Device Please follow the connection diagram above to install the GoIP1 Gateway a Insert a GSM SIM card in the SIM card compartment located at the bottom of the GolP1 Gateway b Connect an Ethernet cable the LAN port of the GolP1 Gateway and the other end to your existing
27. ences rpt is the repeat counter 0 infinite 1 n repeat 1 n times c1on is the cadence one on in milliseconds c1off is the cadence one off in milliseconds c2on is the cadence two on in milliseconds c2off is the cadence two off in milliseconds c3on is the cadence three on in milliseconds c3off is the cadence three off in milliseconds f1 is the tone 1 frequency 300Hz 3000Hz f2 is the tone 2 frequency 300Hz 3000HZ f3 is the tone 3 frequency 300Hz 3000Hz f4 is the tone 4 300Hz 3000HZ p1 is the attenuation index for f1 0 31 0 3dB 1dB increments p2 is the attenuation index for f2 0 31 0 3dB 1dB increments p3 is the attenuation index for f3 0 31 0 3dB 1dB increments p4 is the attenuation index for f4 0 31 0 3dB 1dB increments For example the tone definition for a tone of 450Hz with a cadence of 700 ms on and 1000 ms off is 1 0 700 1000 0 0 0 0 450 0 0 0 20 0 0 0 VAD Innovative 3 3 5 GSM Group Mode One Channel GSM VoIP Gatewa Tiny RK H a E E Ke Zi E a S d E E nn gt d Ss GoIP Host GolP GolP GolP FE i Server Client 1 Client 2 Client N Extension Incoming calls forwarded to GolP clients GolP1 can group multiple devices together and provide line hunt The Group Mode works like a multi channels GSM gateway Any GoIP1 s channel can work as Group Server Mode or Client Mode GSM Group Mode SMS Mode Disable As Server As Glient Server Mode Only one GSM channe
28. field with a new segment address that is different from that on the LAN port Enter the subnet mask in Subnet Mask field accordingly A commonly used value is 255 255 255 0 IP Address 192 168 8 1 Subnet Mask 255 255 255 0 Enable the DHCP Server if you want the GolP1 Gateway functions as a local DHCP host on PC port This will enables the GolP1 Gateway to assign IP Addresses to network devices that are attached to the PC port DHCP Server Enable Disable Starting Address 192 168 8 100 Ending Address Static DNS optional 43 V D Innovative One Channel GSM VoIP Gatewa Specify the Starting Address Ending Address and Static DNS accordingly 4 Advanced The Advanced settings allow the user to set the broadcast address and to clone a MAC address instead of using the factory preset MAC address Please consult your network administrator for further information Advanced lt lt a Address IP Broadcast Address Cd 3 11 Save Configuration To confirm and commit all changes that have been made click on the Save Changes tab Otherwise all changes will be discarded 3 12 Discard Changes To discard all changes made click on the Discard Changes tab 3 13 Tools Menu Select the Tools to access the following functions Online Upgrade Change Password Reset Config and Reboot eem Configurations Last Upgrade Time Current Version GHS 3 01 36 Upgrade site SSCS Tools Online Upgrade
29. l runs in Server Mode The GSM channel that is set in Server Mode will forward the GSM s incoming calls to other available client channels The GSM channel that is set in Server Mode will be your main number for your customer Client Mode Other GSM channels will run in Client Mode The GSM channels that are set in Client Mode will report their status to GSM channel that is set in Server Mode The GSM channel in Server Mode then forwards phone calls to available GSM channels in Client Mode You must enter the GSM number for that GSM channel and IP address of the device in Server Mode into the field ServerAddress 1 esmNumber SS Disable Please set all channels to Disable Mode if you would like to run each channel V M DW Innovative independently One Channel GSM VoIP Gatewa 3 3 6 GSM Caller ID Anonymous 6SM CallerlD Fees O Enable Disable Some GSM ISPs allow the caller to disable the phone number caller ID when making outgoing calls This feature must be supported by GSM ISPs 3 3 7 GSM Band GSM Band 900 1800 w 900 800 85071900 GolP1 Supported quad GSM bands 850MHz 900MHz 1800MHz 1900MHz Select the correct GSM bands that are used in your country 3 3 8 SMS Sender OMS Sender SMS Server IP fo SMS Server Port Po SMS Client ID Po Password oO VADcore offers a software to send out SMS to GSM network through GoIP1 Gateway A SMS server is required to work with GoIP1 Gateway for SMS Sender Please
30. ntered the GolP1 will generate an indication tone and wait for the caller to dial the password 7 Trust List Forward to VoIP Auth Mode PSTN Trust Lists PSTN Trust List Trust Number Po Trust Number2 CE Trust Number ee Trust Number4 Po Trust Number5 Po Trust Numbers ee Enter the phone numbers in the Trust Number field if Trust List is used People calling from the trust phone numbers will be able to use GoIP1 to connect to VoIP 8 Password or Trust List Callers will be able to use GoIP1 for VoIP connection if their phone numbers are on trust phone number list or if they have the password 29 VAD Innovative 3 6 SMS Disposal One Channel GSM VoIP Gatewa 3 6 1 SMS Call Out GoIP1 Gateway supported SMS call In this mode when GoIP1 Gateway receives a SMS from a mobile phone it will automatically make a call to SIP server To use this function select the SMS Dial option in configuration page SMS Mode Disable kl Disable CID Forward Mode SMS Mode SMS Dial SMS Dial Prefix CID Forward Mode A Mode 1 GolP1 dial the call use SMS sender call ID B Mode 2 GolP1 dial the call via its VoIP account and add the SMS sender phone number to Call Divert option s Forward Number VoIP to PSTN automatically C Mode 3 GolP1 dial the call via its VoIP account and add the SMS sender phone number to SIP invites header D SMS Dial Prefix When GolP1 dial a SMS call it will automatically add this
31. one number list or if they have the password One Channel GSM VoIP Gatewa VoIP Trust List lt SIM Card Settings SIMCardExpiry Sid aes SE Report Number fae Report Time sMcadiD CS 1 SIM Card Expiry usage limit minutes 2 SIM Card State Report Number the recipient phone number for the SMS report 3 SIM Card State Report Time the time schedule to send SMS report 4 SIMCard ID Identification sent with the sms message 3 5 2 Call Forward From PSTN To VolP Forward to VoIP Enable Disable lavas EEN PSTN To VoIP VoIP Forward to VoIP Auth Mode No Auth v PSTN Trust List gt Forward Number 28 V D Innovative One Channel GSM VoIP Gatewa Enter a phone number in this field will forward all incoming PSTN GSM calls to this phone number a VoIP number If this field is blank the GolP1 answers all incoming GSM calls and then generates the dial tone The caller can then dial a VolP number When finishing a pound can be dialed to activate the dialing to the VolP number immediately If a pound is not input the VoIP number will be dialed after a preset timeout When Dial Plan Please refer to 3 9 Dial Plan for details If is entered in the field all of the phone calls will pass through Forward to VoIP Auth Mode This field sets protection for using the GSM connection to VoIP 5 No Auth Anyone can make phone calls through GolP1 6 Password If a password is e
32. option s digit in be Called ID Mode 1 examples A GoIP1 use SMS Dial Mode 1 30 VAD Innovative One Channel GSM VoIP Gatewa PSTN Trust List GSM Group Mode SMS Mode Dial sms DialPreix A mobile phone s number is 86 13800000000 it sends a SMS 8675588228822 to GoIP1 s GSM SIM card When GolP1 device receives this SMS it will automatically call the number 8675588228822 and the caller is number 8613800000000 The sent out signaling as follow Sending Message to 192 168 2 1 5060 INVITE sip 8675588228822 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 189 5060 rport branch z9hG4bK1686911003 From lt sip 8613800000000 192 168 2 1 5060 gt user phone tag 626918067 To lt sip 8675588228822 192 168 2 1 gt Call ID 1835068843 192 168 2 189 5060 CSeq 2 INVITE Contact lt sip 8613800000000 192 168 2 189 5060 gt MaxF orwards 30 User Agent HyberTone Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 B GoIP1 use SMS Dial Mode 1 and add a prefix as 999 SMS Mode Dial SMS Dial A mobile phone s number is 86 13800000000 it sends a SMS 8675588228822 to GolIP1 s GSM SIM card When GolP1 device receives this SMS it will automatically call the number 9998675588228822 and the caller is number 8613800000000 The sent out signaling as follow 31
33. or the appropriate IP settings Obtain an IF address automatically Use the following IP address IF address 192 168 8 10 Subnet mask 255 pp 255 0 Set an unused IP address that is in the same segment as the PC port address Once either the IP address of the LAN or PC port is known you are now ready to access the Web server of GolP1 Gateway VADESE 3 1 Web Configuration Menu One Channel GSM VoIP Gatewa If your computer is connected to the GolP1 Gateway via the LAN port you need to type the LAN IP address of the GolP1 Gateway in your Web Browser to access the Web server of the GolP1 Gateway The default IP address on the LAN port is 192 168 0 100 Address not valid Windows Internet Explorer Gg bon 192 Jee 0 1004 File Edit View Favorites Tools Help qF Favorites iG Connecting If your computer is connected to the GolP1 Gateway via the PC port you should type GoIP1 s PC port IP address 192 168 8 1 in the Web Browser Heel Microsoti Internet Epionen File Edit View Favorites Tools Help sax z P DN a A pP Search 5 Favorites E A d IO Address 192 168 8 1 If the connection is correct the Web Browser will prompt you to enter the User name and Password as shown below Enter the User name and Password and the press OK to access the GoIP1 Gateway Web Server The default for both user name and password is admin V A B Innovative One
34. oxy periodically in order to keep the NAT open on your firewall for incoming data traffic 3 4 3 Advanced Timing Advanced Timing lt a 180s NICT Expiry 2 180s ICT Expiry 6 3608 Retransmit T1 200 2000ms Retransmit T2 2000 8000ms DTMF Signaling Outband Outband DTMF type RFC 2833 v RTP Payload Type Signaling QoS None v Signaling NAT Traversal None w Media Settings gt A No Answer Expiry 32 180s NICT Expiry 2 180s Retransmit 1T1 200 2000ms Retransmit T2 2000 8000ms Some SIP proxies may have special timing requirements Change these parameters as required B DTMF Signaling 1 DTMF TYPE DTMF signals can be sent over to the called party after a call is established GolP1 Gateway supports both Inband and Outband DTMF signal types DTMF Signaling Outband wi QOuthand For Inband DTMF type DTMF signals are generated locally at the calling phone and then 21 VAD Innovative One Channel GSM VoIP Gatewa send to the called party as part of the voice signals This method is not reliable since the quality of the DTMF signals is subject to the Codec used and the quality of the networkt For Outband DTMF type DTMF signals are independently translated and sent to the called party After receiving DTMF signals the called party translates and interpret based on the DTMF protocol This method allows more reliable DTMF signaling However it requires the called party to support this featur
35. quential and Fixed The default is set to Fixed mode with the fixed delay of 60ms Please consult your network administrator for more information on the network environment in order to determine the optimal settings 23 VAD Innovative D Media Qos One Channel GSM VoIP Gatewa Media Qos IF TOS DiffSery Similar to the Signaling QoS the Media Qos in intended to improve the voice performance or quality if your local network supports QoS E Media Encryption GolP1 Gateway supports different encryptions for voice media Select the one that you prefer F Symmetric RTP Normally GolP1 Gateway uses RTP ports based on the configuration If this box is checked GoIP1 Gateway will identify RTP ports from the media traffic it has received and use the same ports when sending media traffic G Media NAT Traversal Similar to Signaling NAT Traversal this feature allows media packets RIP to be routed properly in various network environments Media NAT Traversal STUN RFO 3489 For forward DMZ Relay Proxy 1 None Select None to disable this feature 2 STUN RFC 3489 STUN Simple Traversal of UDP User Datagram Protocol through NATs Network Address Translators is a network protocol allowing a client behind a NAT or multiple NATs to find out its public address the type of NAT it is behind and the internet side port associated by the NAT with a particular local port Select STUN RFC 3489 to use a STUN server
36. sans se SonisexentesaaoxsnesSusensa ssiseotonadses 18 34 2 Advanced SCEEINGS esanic aE arn EENT ATRAER EEE 20 3 4 3 Advanced Tmmg EE 21 3 4 4 E ER dr E 23 VAD Innovative One Channel GSM VoIP Gatewa 3 4 9 Codec EE et 25 BO GU DIV E 26 3 5 1 Call Forward From VolP To PSTN ccccccsssscecccceessseeececesesessseeceeseessseseseeeeeeeseseeeees 26 3 5 2 Call Forward From PSTN TO VOIP EE 28 ME ee CN 30 EC CIR RTE 30 OZ SIO CY ca sours sores oon tan E te veonotsis auc san esasesecnadeon E E 34 3 6 2 1 SMS Relay To VoIP System 34 3 6 2 2 SMS Relay To GSM Network EE 35 3 7 GSM Caller ID TramS Par GU vsccssieccececicevececeseceisccesssstaveuscavseeccenscieeeseaesedscisesstansiacaveecteces 36 SEENEN 37 Pe SAIN EE 39 3 10 Network Configuration E 40 Se E Ee 40 3 10 27 PE Port EEGENEN 42 3 11 SAVE Confio U AON E 44 B02 HR Et 44 Bis Wed TOS MG EEN 44 BAe OMS gel 44 313 4 Chala e E 45 3513 3 Reset COMLSULALION ccsssereresovessensacvanssaetcrzsenicctsniasnedinadeosadesaacounseasuestectaseassnacuenentiotsnancenioece 46 3 13 4 Reboot the NU Le EE 46 4 Hardware Specifications ssscccsssssssssecssssecsesssesescscsessesescsessscsessseessseesssceecsees 47 5 Manufactory elef DC 48 VAD Innovative 1 Product Introduction One Channel GSM VoIP Gatewa 1 1 General Information A VoIP GSM Gateway enables direct routing between IP and GSM network without the use of a FXO port or the PSTN network With this device the usag
37. t the GoIP1 reboots itself automatically Note lt password gt is the webpage login password as described in Section 3 1 V Mm DW Innovative 3 Configuration Guide One Channel GSM VoIP Gatewa To configure the GolP1 Gateway you must login to its Web server via the LAN or PC port The LAN port is factory preset to IP address 192 168 0 100 and the PC port is set to the fixed IP 192 168 8 1 If you lose the IP address information for LAN port just dial a call to GolIP1 Gateway s SIM card phone number When the call is connected you will hear a dial tone Then dial 01 for English voice prompt on the LAN IP and 00 for Chinese voice prompt on the LAN IP The LAN IP Address can also be obtained by sending a SMS message to the GSM phone number The GolP1 will then reply with a SMS message containing the LAN IP address If you want to obtain LAN port IP by sending a SMS message please send INFO or info Another way to access the GolP1 Gateway is via the PC port You will need to change your computer s LAN configuration via the Network Connections under the Control Panel Windows Control Panel gt Network Connections gt Local Area Connection Property gt Internet Protocol TCP IP s Property Internet Protocol TCP IP Properties VE General rou can get IF settings assigned automatically if your network supports this capability Otherwise You need to ask your network administrator f
38. uence When Forward Number field has a phone number GolP1 will automatically forward all VoIP calls to this phone number When Forward Number is empty GolP1 will route phone calls based on the following conditions A When the Callee ID is GoIP1 s SIP account number GoIP1 will take the call and feed back a dial tone to VoIP caller Then VoIP caller must dial a PSTN number when hearing this dial tone B When the Callee ID is not GoIP1 s SIP account number GolP1 will automatically dial out with this number thru GSM network based on the rules in Dial Plan VoIP to PSTN field Dial Plan Please refer to 3 9 Dial Plan for details If is entered in the field all of the phone calls will pass through Forward to PSTN Auth Mode This field sets protection for using GoIP1 to connect to GSM network 1 No Auth Anyone can make phone calls through GolP1 2 Password If a password is entered the GolP1 will generate an indication tone and wait for the caller to dial the password 3 Trust List 27 VAD Innovative VoIP Trust List Trust Number Po Trust Number2 PY Trust Number3 Po Trust Number Po Trust Number5 Po Trust Number6 tts Trust Number Ir Enter the phone numbers on the Trust Number field if Trust List is used People calling from the trust phone numbers will be able to use GSM connection 4 Password or Trust List Callers will be able to use GoIP1 for GSM connection if their phone numbers are on trust ph
39. ws the status of Line s connection status If the connection is successful this field displays LOGIN otherwise it displays LOGOUT 3 2 2 Network Information A E LAN Port Configuration This field displays the status of the LAN port PC Port Configuration This field displays the status of the LAN port PPPoE If PPPoE is enabled it displays its status Default Route This field displays the IP address of the default routing Gateway DNS Server This field displays the IP address of the Domain Name Server 3 2 3 GSM Module Information GSM Module This field displays the GSM module type GSM Signal This field displays the GSM signal status The value of GSM signal strength RSSI Received Signal Strength Indication is between 0 dbm and 31 dbm The value of 99 means unknown or undetectable GSM Status This field shows the status of GSM connection status If the connection is successful this 12 Innovative VADE One Channel GSM VoIP Gatewa field displays LOGIN otherwise it displays LOGOUT 3 3 Configurations Click on the Configurations tab on the left hand column to access the device configuration menu Preference Network Call Settings Call Divert Save Changes and Discard Preference Changes Status Configurations Preferences Network Call Settings Call Divert Save Changes Discard Changes Tools Time Zone GMT 8 Time Server pool_ntp org EE Time Gap Remote
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