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GoIP Series SIM Card for GSM Voice Gateway User Manual

1. 69 http www dbltek com 5 Factory Default Parameter Table GoIP GSM Series Voice Gateway http www dbltek com GoIP GSM Series Voice Gateway e Overview 1 1 Introduction The GoIP series gateway is a broadband relay gateway newly developed by DBL Technology It is a new product for seamless connection between the GSM network and VoIP network When the mobile phone SIM card is installed in the GoIP users can register the GSM phone to the VoIP softswitch system Through the GoIP users can realize the uplink and downlink calls between the GSM network and the VoIP network In addition the GoIP supports the transparent transmission of the caller number from the PSTN to the VoIP The GoIP features embedded SIP and H 323 protocols with flexible setting The bi directional password authentication call authorization and trust list authentication greatly minimize the risk of charge losses and the flexible routing function can meet special requirements of various call forwarding In particular the GoIP gateway supports multi device groups with flexible setting of large GSM gateway groups with different channel numbers With its low price excellent voice quality and powerful features the GoIP series gateway is the first choice for system integrators traffic operators and softswitch manufacturers The GoIP series gateway includes GoIP GSM and GoIP_4 EE ET Hm LN Ex g m PSTN Gateway I
2. A No firewall The firewall traversal mechanism is not supported B Port Transparency DMZ The port transparency is used to transfer the network port on the LAN interface to the computer or the server in the LAN This feature enables external users through the Internet in most cases to share the services of internal servers such as FTP HTTP and Telnet The port transparency supports the address of the gateway and response server The gateway is a communication device that connects two different networks The response server is a standard service device that implements the ECHO protocol C STUN RFC 3489 The Simple Traversal of UDP over NAT STUN is a protocol that enables the SIP telephone to detect the existence and type of the firewall installed in the computer This parameter indicates the SIP address of the STUN server Note The STUN protocol supports the SIP gateway only D Trunk Agent The trunk agent protocol is a firewall traversal technology developed by DBL Technology It enables the products of DBL Technology to be applicable for most LANs It involves the address port user name and password The trunk agent protocol supports encryption on communications over the gateway This feature needs the support of the server developed by DBL Technology 44 http www dbltek com GoIP GSM Series Voice Gateway Media agent mode Mode 1 media encryption and agent supported by all versions of relay servers Mode 2
3. 48 9 TI Call ROULE et bbb uisu mad dece ati anta eh muse ua cue d Fula edad ak tu MO ru IO dud au UH UP IR ad 48 a T 2 AMINE NICATION Mode Se llle oi rod Poco ee E neo e cuiu obe oa itn odd fases 49 3 7 2 1 Password Authentication ccccccccceeeeeeeeeesseeseeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeees 49 IRZ 21 CUS E Lr AVL acit O ass oestusi Antaina tse tarl deisel e e 50 3 7 2 3 Password or Trust List Authentication cccccccccccecccceeeceeesesseeseeeeeeeeeees 51 3 9 alc D EEOTIO DIC EEE i EET E T ENT EET Lia ia Le Le Li Le 51 2 9 S NDS AVTOGIS Lesicaieostebtscsioentudse emo one adero eid Ut oisi Utt cii LEAL ini eR en 53 2 9 TS NES Dialing nder SIP Protocol Soie eee utei a deed ufa hte leve e ee qt asieneedaake 53 3 9 2 SMS Dialing under the H 323 Protocol acetone etosu sucks qud oer n e ehe ede 58 IIG S NISSDODWHEOITIE Sos esee ciat Cri monto Dto zip Sa Ud Arete evn Te ay Te 63 3 10 Transparent Transmission of PSTN Caller Numbers eeeeeeee 64 S DI sSavesthexc halle aaa ea ae aaa 66 JA Abandon the Cane NORTE A 66 SJ TO0l sab ute ML uL ML ML ML M AU IL M E e 67 Sol Bi Onlus DSPAOIIS 2 haces Strat Sasha ded a a urit 67 3241572 ModiHcatonof PASS WOE cosi mt ora re oc e e dise Ra ek pedi etoa pb Sa eoe EON asap 67 2 15 23 Restore Factory SCUIN GS mosini or besote uve pues ane nemo owes ERE E 68 MEA De Se ee RM ae RE aia ee aa ee a TUM 68 4 Parameters of Equipment
4. Mode 1 When a short message of 8675588228822 is sent from the mobile phone number 86 13800000000 to the GoIP the GoIP will send the following call request signaling 54 http www dbltek com GoIP GSM Series Voice Gateway 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 2 237 5060 branch z9hG4bK363969813 From sip 8613800000000 192 168 2 1 5060 gt user phone tag 65248630 To lt sip 8675588228822 192 168 2 1 Call ID 117025903 192 168 2 237 CSeq 2 INVITE Contact sip 8613800000000 192 168 2 237 5060 gt Max Forwards 30 User Agent DBL Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 When the user enters an SMS dial prefix such as 999 the above call request signaling is changed to SMS Mode Dial SHS Dial Mode 1 Sending Message to 192 168 2 1 5060 INVITE sip 9998675588228822 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 237 5060 branch z9hG4bK363969813 From lt sip 8613800000000 192 168 2 1 5060 gt user phone tag 65248630 To lt sip 9998675588228822 192 168 2 1 gt Call ID 117025903 192 168 2 237 25 http www dbltek com GoIP GSM Series Voice Gateway CSeq 2 INVITE Contact lt sip 8613800000000 192 168 2 237 5060 gt Max Forwards 30 User Agent DBL A
5. media encryption and agent supporting the transit over the specified port supported by V2 relay server Mode 3 media encryption and agent for the conversion of RTP data to TCP packet supporting the transit over the specified port Supported by the relay server later than V2 3 5 7 Dialing Rule The GoIP supports number dialing by rules You can specify dialing rules in the dialing rule parameter of the Call Forwarding 3 5 7 1 Format of Dialing Rules 1 You can specify multiple rules which are separated by the delimiter l For example 00 0010 04 861 86755 2 The number is matched from the left of the dialing rules to the right When the number matches the correct rule the number stops matching Otherwise the number continues to match the next rule 3 The rule format is AA aa bb such as 0 0 86 Where AA indicates the number to match and aa bb indicates detailed actions to be taken on the number If the number is successfully matched aa is deducted and bb is added If the number fails to match the number continues to match the next rule If no digit after the colon is specified such as 00 it indicates that no actions are taken when 00 is matched and the number exits the matching If no digit before the colon is specified such as 86755 it indicates that instead of matching actions are taken on the number directly 4 You can specify a range
6. Anonymous Serve as the client When GoIP operates in this mode it will send its real time status to the server of the GoIP group so that the GoIP server can deploy the call forwarding Network Tones China GSM Group Mode asClient v Server Address o GSM Number o Server address It is the IP address of the GoIP gateway of the GoIP group server GSM number It is the telephone number of the GSM SIM card used by the GoIP 3 3 6 Anonymity of the GSM Caller Number nnd C Enable Disable GSM Band anon aon M Auto Reboot Reboot Time 04 00 China Phone Code Enable Disable Iv MR The caller number can be hidden but this needs the support of the GSM operator 3 3 7 IMEI IMEI 359094025200031 The IMEI International Mobile Equipment Identity is an electric serial number containing 15 digits 19 http www dbltek com GoIP GSM Series Voice Gateway 3 3 8 SMS Send to Client SMS Sender Enable Disable sMsSeweriIP SMS Server Pot SMS Client ID pe The SMS server sends the request to the mobile phone via the GoIP or the mobile phone sends the request to the SMS Server via GoIP SMS Sender Enable Disable SMS Server IP 182 158 2 2 SMS Server Port 44444 SMS Client ID SMS server address This is for filling the IP of the SMS server Please make sure that server is installed with the GoIP SMS management server software independently developed by DBL Techn
7. This parameter is used to display the name of the user who subscribes the H 323 service For example when you call your friend John Smith your name will be displayed on your friend s telephone 27 http www dbltek com GoIP GSM Series Voice Gateway C H 323 ID H232 ID is used to verify the account Users can set this parameter according to the requirements of the service provider D Default Voice Gateway This parameter is used to find the proper Gatekeeper or IP address of the equipment of the callee Enter the IP address such as 192 168 2 197 or domain name such as gk yourisp com If the softswitch system uses the non standard port 1719 and 1720 you can add the detailed port number at the end of the IP address or domain name of the Gatekeeper For example if the port number is 7300 the IP address is 192 168 2 197 7300 and the domain name is gk yourisp com 7300 Under the direct connection mode the GoIP will forward all calls to the VoIP network to this address Note The value of this parameter must be standard ASCII characters enter characters under the English input mode 3 5 1 2 Gatekeeper Endpoint Type H 323 Phone Advanced Settings Endpoint Mode Gatekeeper Wade Media Settings Config Mode Single Config GaeWayPre a Display Name a i Gatekeeper Address Enable VOS AVS Signaling Encryption Enable Authentication 1 Under the Gatekeeper mode the GoIP operates in the H 323
8. call voices together without any processing Therefore the inband DTMF transmits DTMF signals through a single way 2 Outband DTMF The outband DTMF transmits dialing tones over protocols such as RFC2833 and SIP INFO which can ensure the validity of the transmission G Registration Mode When the registration information is sent to the platform under Mode 1 experise info is included When the registration information is sent to the platform under Mode 2 experise variable is not sent 3 5 4 Media Advance Setting The media advance setting is set for the RTP media stream of the gateway Select Call Setting gt Media and the following setting parameters are displayed Advanced Settings Media Settings RTP Port Range PacketLength ms Jitter Buffer Fixed o Delayims Media QoS None Media Encryption None i RY Symmetric RTP Media NAT Traversal Mone Audio Codec Preference A RTP Port Range This parameter is used to specify the UDP of the RTP and used with the router port for mapping Note The terminal will use multiple pairs of RTPs depending on the number of lines that the terminal supports The value of this parameter ranges from 5500 to 5520 40 http www dbltek com GoIP GSM Series Voice Gateway B Packet Length ms This parameter indicates the duration of sending a network packet If this parameter is null it indicates that the default value is 20ms I
9. can add the detailed port number at the end of the IP address or domain name of the proxy server For example 192 168 2 26 3000 or hy con com 3000 D SIP Registration Server The SIP login server is a server that the gateway registers the account This parameter is used to set the IP address or domain name of the SIP registration server If the SIP registration server uses the special port other than the SIP default port 5060 you can add the detailed port number at the end of the IP address or domain name of the registration server For example 192 168 2 26 3000 or hy con com 3000 E Outbound Proxy The outbound proxy is mainly used in the scenarios where the firewall or NAT exists so that the signaling and media stream can penetrate the firewall F Homing Domain This parameter is used for the domain management host of the SIP a host that provides the SIP service G Authentication ID This parameter is used to set the authentication account when the gateway logs into the SIP proxy server H Password This parameter is used to set the authentication password when the gateway logs into the SIP proxy server H Display Name When you call your friend John Smith your name will be displayed on your friend s telephone J Backup Server 36 http www dbltek com GoIP GSM Series Voice Gateway Backup Server Enable Disable aero E cc Registrar Server me i LL Domain This parameter is used for
10. com 3 3 11 Prompt Tone System GoIP GSM Series Voice Gateway The prompt tones are the combination of the intervals and frequencies of the dialing tones and ring back tones when users hook off the telephone You can select the following prompt tones for the GoIP according to the countries and regions where the GoIP is used so as to remain conventional tones Network Tones GSM CallerlD Anonymous GSM Band Reboot Time China Phone Code Customize China Australia 0000 Crechosloyakia Germany Hong Kang Korea Mew zealand Slovenia United Kingdom United States Customized Users can customize prompt tones according to their special requirements Select Customize the following setting parameters are displayed Network Tones Dial Tone Ring Back Tone Busy Tone Indication Tone Customized The setting parameters are defined as follows Each prompt tone involves the following parameters If a parameter is not defined the value of the parameter shall be 0 nc rpt clon cloff c2on c2off c3on c3off f1 f2 3 f4 pl p2 p3 p4 gt nc the number of tones 1 4 22 http www dbltek com GoIP GSM Series Voice Gateway rpt the number of repeats 0 to infinity clon the duration when the frequency 1 is on ms cloff the duration when the frequency 1 is off ms c2on the duration when the frequency 2 is on ms c2off the duration when the frequency 2 is of
11. mobile phone numbers 130xxxxxxxx to 130xxxxxxxx and telephone number 1xxxxxxx to 8xxxxxxx respectively Note When the length of the number is specified the exceeded numbers will be discarded if the length of the number exceeds the specified length For example The rule is 0 113 0 9 xxxxxxxx Ol 1 8 xxxxxxx 0755 When you dial the number 88990011 and 8899001133 the result is the same The number actually dialed is 075588990011 3 6 Volume Adjustment The GoIP has a built in volume adjustment panel which should be used carefully When you need to adjust the volume of the gateway change the address http XXX Xxx xxx xxx xxx gain html to http Xxx xxx xxx xxx xxx gain html Then the following volume setting page is displayed 47 http www dbltek com GoIP GSM Series Voice Gateway Gain Settings Line 1 Line 1 Output Gain loo Line 1 Input Gain 3 Line 2 Line 2 Output Gain T Line 2 Input Gain Line 3 Line 3 Output Gain Line 3 Input Gain T Line 4 Line 4 Output Gain T Line 4 Input Gain T After the volume setting is completed click Save and the setting will take effect immediately for ongoing calls the setting will not take effect immediately Note The adjustment on the output volume of the line may cause the terminal to fail in dialing numbers Therefore set this parameter carefully The adjustment on the input output volume is for VoIP lines 3 7 Cal
12. or hy con com 3000 D Outbound Proxy The outbound proxy is mainly used in the scenarios where the firewall or NAT exists so that the signaling and media stream can penetrate the firewall E Homing Domain This parameter is used for the domain management host of the SIP a host that provides the SIP service F Authentication ID This parameter is used to set the authentication account when the gateway logs into the SIP proxy server G Password This parameter is used to set the authentication password when the gateway logs into the SIP proxy server H Display Name When you call your friend John Smith your name will be displayed on your friend s telephone I Backup Server Backup Server Enable Disable paisa Ge b Registrar Server M q Domain This parameter is used for registration backup When a backup registration server exists in the user s system the user can enable this parameter Once the backup server is enabled the gateway will automatically log into the backup server in case of the failure of the main Server 34 http www dbltek com GoIP GSM Series Voice Gateway 3 5 2 3 Setting by Line Only Valid for the GoIP_4 Call Settings Endpoint Type SIP Phone Advanced Settings Config Mode Config byLine gt Media Settings gt Linet Line Line 3 C Lined Phone Number 20 Display Name Gateway Prefix SIP Proxy 192 158 2 7 SIP Registrar Server 1
13. port It is used for communications between the SIP agent and the SIP proxy server as well as other SIP managers B NAT Hold This parameter is used to hold the port that is activated by the NAT for SIP signaling communication The unit of the parameter is m NAT Keep alive C Enable Disable C Timeout Setting Advanced Timingss boi T 32 1805 oo U00ms sat lla a un ms Su E Signaling QoS Quality of Service QoS is a network s capacity to provide priority services including the special bandwidth jitter control and delay used for real time and interactive traffic and improvement of the packet loss ratio This parameter is used to mark the specified QoS label for the call signaling packet to increase the network service quality Signaling Gos Signaling Encryption Ps Signaling HAT Traversal F DTMF Signals DTMF signals are used to transmit call signals to the call switching center over the audio band The DTMF means that two different frequencies of sounds are combined into 16 types of dialing tones The telecom office or 1860 service hotline identifies these dialing tones through analyzing the DSP and thus determines the dialing number There are two types of DTMF signals inband DTMF and outband DTMF 39 http www dbltek com GoIP GSM Series Voice Gateway DTMF Signaling Inkang bd Signaling QoS Cee ee i halan 1 Inband DTMF The inband DTMF transmits dialing tones and
14. www dbltek com GoIP GSM Series Voice Gateway 3 2 1 Telephone Information A Product Sequence Number Each GoIP gateway has a factory set sequence number such as GOIP08030031 which is used for centralized setting technical support and maintenance filing The sequence number is printed on the bottom plate of the gateway and is read only B Software Version It displays the current version of software used by the GoIP When you want to upgrade the software make sure the update version is newer than the current version C Hardware Version It displays the current hardware version of the gateway D Line Register Status It displays the login status of the line When the line has logged into the SIP server or H 323 Gatekeeper LOGIN is displayed otherwise LOGOUT is displayed E Line Use Status It displays the use status of the line When the line is in use the status is ACTIVE when the line 1s idle the status is IDLE 3 2 2 Network Information A LAN Port It displays the current IP address of the LAN port such as 192 168 2 172 B PC Port It displays the current IP address of the PC port C PPPoE dialing It displays the PPPoE broadband connection condition After the connection the IP address obtained is displayed on the LAN port D Default Route 14 http www dbltek com It displays the current gateway address E Domain Name Server DNS It displays the current DNS address 3 3 User Options G
15. you call your friend John Smith your name will be displayed on your friend s telephone D H 323 ID H232 ID is used to verify the account You can enter this parameter according to the requirements of the service provider E Gatekeeper Address This parameter is used to find the proper Gatekeeper Enter the IP address of the Gatekeeper such as 192 168 2 197 or domain name such as gk yourisp com If the softswitch system uses the non standard port 1719 and 1720 you can add the detailed port number at the end of the IP address or domain name of the Gatekeeper For example if the port number is 7300 the IP address is 192 168 2 197 7300 and the domain name is gk yourisp com 7300 Under the Gatekeeper mode the GoIP will forward all calls to the VoIP network to this address Note The value of this parameter must be standard ASCII characters F Enable VOS AVS Encryption You can enable the VOS AVS encryption 29 http www dbltek com GoIP GSM Series Voice Gateway Iv Enable vOSIAVS Signaling Encryptian Encryption Mode vog Signaling Encre VOS Signaling And Marliz AWS Signaling Encrvptia AWS Signaling And signa G Enable Authentication Auth Click Enable Authentication and enter the following parameters when you need to set the H 235 authentication code and password Iv Enable Authentication 3 5 1 3 Advance Setting of the H 323 The advance option of the GoIP involves the signaling and media
16. 202 96 136 145 update GHS 4 01 12 pkg and then click Start The gateway begins to upgrade After the GoIP is successfully upgraded the gateway will automatically restart Online Upgrade Last Upgrade Time Current Version GHS 4 01 12 Upgrade Site 118 142 51 182 update GHS 4 01 12 pkg Note During the upgrading do not cut off the power Otherwise the GoIP will be damaged 3 13 2 Modification of Password You can modify the password of the user and administrator Select Tool Modify Password The password modification page is displayed as shown in the following figure Enter a new password and click Change Then the password is successfully modified 67 http www dbltek com GoIP GSM Series Voice Gateway Administration Level Note The password modified by users will be cleared and restored to the factory default password after the factory settings are restored 3 13 3 Restore Factory Settings Select Tool gt Restore Factory Settings The following prompt is displayed Microsoft Internet Explorer X 1 J Are vou sure to reset to Factory default Click OK All the parameters of the gateway will be cleared and the gateway will automatically restart After the gateway is restarted all the settings restore to the factory default settings This feature can be completed by using the command For details about this operation see the section of Instructions 3 13 4 Reset Configurations Ne
17. 92 158 2 7 Register Expiry s Outbound Proxy Dil Home Domain Authentication ID 120 Password Call Forward Type Mot Forward M Call Forward Humber Backup Server C Enable Disable You need to set parameters for each line and the setting method is the same The setting parameters are as follows A Telephone Number This parameter is used to set the telephone number of the line The telephone number is an unique ID when the gateway serves as the callee B Gateway Prefix The gateway prefix enables the connection of calls through a particular line It can match the first digit only You can set a gateway prefix for multiple lines When you set a gateway prefix for multiple lines the calls that have the same gateway prefix will select the line set with this gateway prefix For example the gateway prefix is 1 When the user dials 10086 the call will be connected by the line with the gateway prefix 1 When the user dials 075588290211 the system detects whether the line with gateway prefix O exists If exists the call will be connected Otherwise the call will be released Note When you set the GoIP by lines the gateway prefix must be set Otherwise the call will not be connected 35 http www dbltek com GoIP GSM Series Voice Gateway C SIP Proxy Server This parameter is used to set the address of the SIP proxy server If the SIP proxy server uses the special port other than the SIP default port 5060 you
18. A DC transformer 3 An Ethernet cable 2 m 1 6 Product Appearance http www dbltek com GoIP GSM Series Voice Gateway 1 LAN The network input port that is connected to the router Modem and switch 2 PC The network output port that is connected to network sharing equipment less than 100 terminals 3 12V 2A DC The output terminal that connects the transformer equipped with delivery 4 Reset The reset switch for quick restart of the GoIP 2 Installation 2 1 Installation Procedure The GoIP has 1 4 SIM card slots an LAN port and a PC port The installation procedure is as follows Open the bottom cover of the GoIP and insert an SIM card of the local GSM network Connect the LAN port with the upper layer network equipment with the Ethernet cable PC port supports network sharing so connect the PC port to the computer or lower layer switch HUB or router 4 Connect the output terminal of the transformer with the power port http www dbltek com 2 2 Connection Figure Plug to Internet internet ADSL Cable Modem Ethernet GSM Network Y t L F 1 A 7 T n Es Channel4 Channel3 Channel Channeli Plug to Internet Internet DASL Cable Modem Ethernet GoIP GSM Series Voice Gateway http www dbltek com GoIP GSM Series Voice Gateway 2 3 LED Indicators The description of LED indicators is as follows This indicator is constantly ON after connected with power m This i
19. Bridging When the PC port is set to the bridging mode the relation between the LAN port and the PC port is layer 2 switching The network equipment connected with the PC port same as the connection with the LAN port B Fixed IP Address 25 http www dbltek com GoIP GSM Series Voice Gateway Select the fixed IP and the following setting parameters are displayed Enter the IP address and subnet mask the network section of the IP address should be different from that of the LAN port to prevent conflict PC Port Static IP IP Address 192 168 5 1 Subnet Mask 2959 299 255 0 DHCP Server Enable Disable Starting Address S Ending Address nl StaticDNS optiona O O Advanced C Enable the DHCP Service This service can be enabled only when the PC port of the GoIP is set as the fixed IP To enable the DHCP service you need to enter the start address and end address E Advance Click Advance and the Hardware address and Broadcast address are displayed in the page The hardware address is used to enter the MAC address in the format of XX XX XX XX XX XX The broadcast address is used to communicate with other computers that are connected to the ATA 3 4 3 Main DNS The DNS domain name system is a database that stores the Internet names and addresses and converts between the name and the common Internet protocol digits The main DNS is the IP address of the main DNS such as 202 67 156 221 or obtain from
20. GoIP Series SIM Card for GSM Voice Gateway User Manual C asa gt Qum a 3 T s ii a AO un E F Pu M e E P d 2i a ud a D f Sw s wu ue bu V3 0 Shenzhen DBL Technology Co Ltd Http www dbltek com Marketing dbltek com Support dbltek com 2010 06 01 http www dbltek com GoIP GSM Series Voice Gateway Content CONTO e 1 Berat 4 TD Tit OGUCH OM oo eo Cette Inter da itat desta diiit att tiab patti 4 1 2 PROLOG ONS Loue oor E i oet dep api Oei eate EN ap ne ditte ot t REMO etn 5 I SH arelwabesEedEULE uns iaodieohi oed oae E arene sate oed tette ee anand teint 2 TA SOP warte Ee ALULG 2o tetro cut Schoo tediae Idest beso Qut cesa co testae Dn aaa E Uo coD o Eos ae S Lo ote teat 5 1 5 Product Package TAS en A N E O HEROS ONLUS 6 1O Product Appearance Socr oem Apad poti iot emot octo i eM AE MINUM eae 6 2 Stala 7 2 T salon PrOCe QUEE naaa a a a orit dee us 7 2 2 COMMECHON PISut iiit odis t ov t et EM Ben a a ee a ei ence eae 8 2 9 Mest DNIe ALOE Sore oet ete E ertt tate CoL eroe erc 9 2d S NIS EnSEEUCELOTIS oai ota tee eue dat eoru eae boe seht Pole das edes aat soon tu e sut Foto du esu au E ooa Mal Eee 11 3 PAGS SEINO eonna A 11 KEPE S ERN etl EET E E E E eR 11 uP OUAlUS adegit dat esM Ld M ECIAM mA LI ELM C LAE ace pete uM S cS MUR E 13 9 22 Telephone InfOrmatioflecuocaiadutren Ge eren ea etn ee dn eti trot Pontio 14 3 222 Network Informations
21. IM NOT INSERTED GSM Operator GSM Signal 21 GSM Status LOGOUT SIM Remain o LIMIT Time Status Configurations Tools Phone Information Serial Number Firmware 65540142 Version Hardware Model GolPx4 Line Register 1 Status LOGIN Line 1State IDLE Line Register 2 Status LOGIN Line 2 State IDLE Line Register 3 Status LOGIN Line 3 State IDLE Line Register 4 Status LOGIN Line 4 State IDLE Network Information LAN Port 192 168 2 226 LAN MAC PC Port 182 158 8 1 PPPoE Disabled Default Route 192 168 2 253 DNS Server 202 96 134 133 GoIP 4 Status Interface GSM Module Information GSM1 Model MTK2 GSM1 SIM NOT INSERTED GSM1 Signal 21 GSM1 Status LOGOUT GSM1 SIM Remain Time GSM1 Number NO LIMIT GSM2 Model MTK2 GSM2 SIM NOT INSERTED GSM 2 Signal 21 GSM2 Status LOGOUT GSM2 SIM Remain Time GSM2 Number GSM3 Model MTK2 GSM3 SIM NOT INSERTED NO LIMIT GSM3 Signal 22 GSM3 Status LOGOUT The GoIP gateway adopts the tree structure The menu is on the left and the setting parameters are on the right as shown in the above figure You can also access the setting page of the GoIP through the IP address 192 168 2 216 or 192 168 2 172 of the LAN port of the gateway The login method is the same as that of the PC port but you must first obtain the IP address of the LAN port 3 2 Status The status page contains the following contents as shown in the above figure 13 http
22. IP DHCP This setting is default If the network for the user provides the DHCP service the GoIP will require the network information such as IP address from the DHCP server automatically B Fixed IP Select fixed IP and the following setting page is displayed Network Confiquration LAN Port IP Address subnet Mask optional Default Route Primary DNS Secondary DNS optional 53 Set these parameters according to the network the user uses 24 http www dbltek com GoIP GSM Series Voice Gateway C PPPOE PPPoE Point to point protocol over Ethernet is a network protocol that compresses the PPP in the Ethernet Select PPPoE dialing and enter the account and password provided by the network provider Network Configuration LAN Port PPPoE D 802 1q VLAN When the network serving the user provides the VLAN service enter the parameter as required 802 10 VLAN Enable C Disable VLAN 1 mos 7 E Advance Click Advance and the Hardware address and Broadcast address are displayed in the page The hardware address 1s used to enter the MAC address in the format of XX XX XX XX XX XX The broadcast address is used to communicate with other computers connected to the GoIP 3 4 2 PC Port Setting The PC port can be set to connect other network equipment through the route or bridging mode The two setting modes are as follows A
23. P 192 168 2 180 5060 branch z9hG4bK1645487913 From sip 2000167192 168 2 1 5060 user phone tag 406202416 To sip 5000 192 168 2 1 Call ID 8472302781 92 168 2 180 CSeq 2 INVITE Contact lt sip 2000 192 168 2 180 5060 gt Max Forwards 30 User Agent HBT Remote Party ID 13800000000 lt sip 13800000000 192 168 2 1 gt party calling screen no privacy oft Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application s dp Content Length 226 Use CID as SIP caller number The GoIP directly initiates the call request to the VoIP system through the PSTN caller number and adds the relevant information to Remote Party ID option of the request signaling The call request signaling is as follows 65 http www dbltek com GoIP GSM Series Voice Gateway 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 z9hG4 bk 1450498491 From 13800000000 lt sip 13800000000 192 168 2 1 5060 gt tag 232569343 To sip 5000192 168 2 17 Call ID 18530689860192 168 2 180 CSeq 2 IMVITE Contact lt sip 13800000000 192 168 2 180 5060 gt Max Forwards 30 User Agent HBT Remote Party ID 13800000000 sip 138000000001 92 168 2 17 party calling screenzno privacy off Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER MESSAGE INFO SUBSCRIBE Content Type applica
24. SMS dialing examples the H 323 number of the GoIP is set as follows Endpoint Tyne 0 323 Phone Advanced Settings Endpoint Mode Gatekeeper Made Media Settings Phone Humber 20001 Gateway Prets DOO pisplayName OOOO H 323 ID 20001 Gatekeeper Address 192 158 2 1 Enable VOS AVS Signaling Encryption Enable Authentication Mode 2 SMS Mode Dial SMS Dial Mode 2 When a short message of 8675588228822 is sent from the mobile phone number 486 13800000000 to the GoIP the GoIP will send the following call requests When the GoIP sends a call request through the H 323 number of the GoIP the GoIP will automatically add the number of the short message sender to the PSTN Forwarding Number in Call Forwarding VoIP Incoming Call Forwarding to the PSTN Immediately In this mode when the GoIP receives the call from the H 323 GK the GoIP will forward the call to the short message sending equipment through the GSM network The call request signaling in this mode is as follows 59 http www dbltek com GoIP GSM Series Voice Gateway Send RAS Message admissionRequest admissionRequest requestSeqNum 241 callType pointToPoint NULL endpointIdentifier 3705 endp destinationInfo 1 elements 0 dialedDigits 8675588228822 srcInfo 2 elements 0 dialedDigits 20001 1 h323 ID 20001 srcCallSignalAddress ipAddress ip 4 octets c0 a8 02 ed port 2049 band
25. Width 2048 callReference Value 7502 conferenceID 16 octets 7f f3 78 77 493f4ccl 9adc6a84 12 d8 30 8f XWI L j 0 activeMC FALSE 60 http www dbltek com GoIP GSM Series Voice Gateway answerCall FALSE canMapAlias FALSE callIdentifier guid 16 octets cb 40 a4 af 8e9b6096 6b5fa003 f2 ed 55 5b O seen Kase gatekeeperlIdentifier GnuGk willSupplyUUIEs FALSE U When the user enters an SMS dial prefix such as 999 the above call request signaling is changed to SMS Mode Dial SMS Dial Mode 2 Send RAS Message admissionRequest admissionRequest requestSeqNum 241 callType pointToPoint NULL endpointIdentifier 3705 endp destinationInfo 1 elements 0 dialedDigits 9998675588228822 srcInfo 2 elements 0 dialedDigits 20001 61 http www dbltek com GoIP GSM Series Voice Gateway 1 h323 ID 20001 srcCallSignalAddress ipAddress ip 4 octets c0 a8 02 ed port 2049 band Width 2048 callReference Value 7502 conferenceID 16 octets 7 137877 493f4ccl 9adc6a84 12 d8 30 8f XWI L j 0 activeMC FALSE answerCall FALSE canMapAlias FALSE callIdentifier guid 16 octets cb 40 a4 af 8e9b6096 6b5fa003 f2 ed 55 5b pO vey ue uus UI gatekeeperlIdentifier GnuGk willSupplyUUIEs FALSE 62 http www dbltek com GoIP GSM Series Voice Gateway 3 9 3 SMS Forwarding The GoIP suppo
26. advance option of the call setting the signaling and media have separate firewall setting as shown in the following figures 3 5 6 1 Traversal of H323 Signaling over NAT The traversal of H323 signaling over NAT firewall is classified into 4 categories Media NAT Traversal Mone STUNIREC 3489 Fort farward D bi Relay Proxy A No The mechanism of firewall traversal is not supported B Nat Citron The Citron is a special firewall traversal protocol for GnuGK and used with GnuGK C Port Transparency DMZ The port transparency is used to transfer the network port on the LAN interface to the computer or the server in the LAN This feature enables external users through the Internet in most cases to share the services of internal servers such as FTP HTTP and Telnet signaling NAT Portfonward DMz v Traversal HAT Address Auto Detect The port transparency supports the address of the gateway and response server The gateway is a communication device that connects two different networks The response server is a standard service device that implements the ECHO protocol D Trunk Agent The trunk agent protocol is a firewall traversal technology developed by DBL Technology It enables the products of DBL Technology to be applicable for most LANs It involves the address of the trunk proxy server port user name and password 42 http www dbltek com GoIP GSM Series Voice Gateway a a NN Encrypti
27. corresponding to Advance Setting and Media respectively Select Advance Setting under H323 and the following setting page is displayed Advanced Settings RAS Port ee Fast Start Enable Disable Fast Start Extend H245 Tunnel aun vl Register Mode Register Multiple Nur gt DTMF Signaling outbang gt Signaling QoS None wm ne Media Settings A RAS Port The RAS is the communication protocol between the terminal and the Gatekeeper It is used to transmit the registration information login broadband change and the status between two 30 http www dbltek com GoIP GSM Series Voice Gateway H 323 unites The RAS port can be used to specify the UDP and used with the router port for mapping B Call Signaling Port Q 931 Port H 225 Q 931 is a call control protocol of the H 323 for transmitting the call setting and unloading information between two H 323 units It is used to specify the Q 931 port TCP that receives calls and used with the router port for mapping C Media Control Port H 245 Port H 245 is the media control protocol of the H 323 It is used to specify the port that receives the H 245 connection TCP and used with the router port for mapping D Fast Start You can enable or disable the fast start described in the H255 0 protocol This parameter is used to detect and solve the compatibility problem If you are not sure do not set this parameter E Fast Sta
28. e language of the setting page when logged in next time 3 3 2 Time Zone and Time Server This item displays the adjusted time according to the selected time zone The gateway receive time and date information from the server through the Network Time Protocol and the time difference will be automatically adjusted For example the pacific standard time PST is GMT 8 and the pacific daylight time PDT is GMT 7 Time Zone GMT 8 Time Server gaal ntp ardg The time zone indicates the zone where the gateway is used You need to enter the correct time zone so that the time of the caller ID and charging information can be displayed correctly The time server is the address of the server that obtains the network time through the Internet The default time server is timekeeper isi edu 3 3 3 DTMF Minimum Detection Interval This parameter is used to set the minimum interval of two DTMF signals Packets may be lost during the data transmission over the GSM As a result a DTMF may be incorrectly identified as two or multiple identical DIMFs when detected by the GoIP The problem of repeated code can be solved effectively through the modification of the parameter cc E Time Gap 16 http www dbltek com GoIP GSM Series Voice Gateway This parameter value ranges from 60ms to 120ms and the default value is 80ms When the value of this parameter is increased properly the repeated DTMF can be avoided efficiently However the packet loss ma
29. e uetus 27 33 1 Direct nei nE E TE 27 pod 7 Gate KCe DET E E E E E E E E EET T TAT 28 Sol Advance setima of NE TES 2 3 eraai a ode eate 30 SNMP uus ce 52 3 9 2 ESNE Mod susto rtd uta eti caet esed bm aput spe usce ede blc ee uu Eme aulet eub ado Fou C Rena 32 0415 2 2 9 1m e TE Se EVE TF MOUSE toa dea usa rd pu ae 33 3 5 2 3 Setting by Line Only Valid for the GoIP 4 eeeesesesss 35 3 9 2 4 Tnk Gateway MO dose Cer pipe edo debate tn ha Dou attese dites 37 See OTP ACV ANC eo TUI PT c E 38 JIA Media Advance e MIC aiie a be id Rae iow iind labiis Bi add 40 3 9 0 VOICE Codines ANd SEQUENCE sicuti restante dew aise Lats e tt utn Du du Dated 4 3 5 0 EXEC W MP onena UO eie Curie aoo Meade Pes coude di edades die cou dest A 42 3 5 6 1 Traversal of H323 Signaling over NAT eeeeeeeeeeem 42 3 5 6 2 Traversal of SIP Signaling over NAT oisi b E rne Ere rt 43 3 5 6 3 Media NAT Traversal sssssssssssssssesssssssssssssssssssssssssssssssssssssessssseees 44 So UIDI Iul e sosideuds ubuntu eaea aaa a 45 3 5 7 T Pormator Dialis RUES ceresna see E ue qug sre aRS LER REN E pk a ERE GU G E RUNE UN ai 45 3 5 7 2 Dialing Rule with Specified Length of Numbers ccccseseeeeeeeeeeeees 46 23 60 Vol tne Ad US MMC ME 20 ia tins a a S M su aa Mit eden bod ion Leda Eau E tiu 47 3 7 Call Forwarding Setting on the Call Route and Authentication Mode
30. elay used for real time and interactive traffic and improvement of the packet loss ratio This parameter is used to mark the specified QoS label for the call signaling packet to increase the network service quality Signaling QoS Mone Signaling NAT None Traversal 3 5 2 SIP Phone The SIP Session Initiation Protocol is a simple network protocol that has less hierarchy and facilitates the initiation of calls among users The calls may be conducted between two or more users which include the sounds images session interactive games and virtual reality 3 5 2 1 Setting Mode The VoIP channel of the GoIP can be set in the following three modes single server line setting and trunk gateway Endpoint Type SIP Phone Contig Mode Single Server Made Phone Number Single Server Mode Config by Line Display Name Trunk Gateway Made Figure 3 31 Setting Mode in the SIP Terminal 32 http www dbltek com GoIP GSM Series Voice Gateway A Single server mode Multiple VoIP channels can share the same setting B Line setting Each VoIP channel can be served by different service providers or served by the same service provider In the latter case multiple different telephone numbers accounts can be registered on the same service so that each telephone number is bound to the corresponding VoIP channel C Trunk Gateway This mode is used to establish the connection or channel between the softswitch and the gat
31. ess of the PC port and set the IP of the computer to dynamic IP or fixed IP as 192 168 8 xxx and the default gateway as 192 168 8 1 A Hypertone VoIP enmnal E Mierozofe lnitorioe Exolorsr File Edit View Favorites Tools Help Q x M gt x a A J Search jg Favorites 4 2 To lo Address 192 168 8 1 Open the Internet Explorer and enter 192 168 8 1 or http 192 168 8 1 in the address bar then the login page is popped up for password input Enter the login account admin as default in the User Name and password admin as default in the Password field The server 192 168 8 1 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 User name v Remember my password Click OK button and the gateway status page is displayed as default 12 http www dbltek com GoIP GSM Series Voice Gateway Status Configurations Tools Phone Information Serial Number GOIPM10051082 Pwera 6149 9 01 30 11 Version Hardware Model eid Phone Status LOGOUT Network Information LAN Port 192 168 2216 LANMAC 00 11 BE 03 4F C8 PC Port 192 158 8 1 PPPoE Disabled Default Route 192 168 2 253 DNS Server 202 968 134 133 VPN Status Disconnect GolP Status Interface GSM Module Information GSM Model MTK2 GSM S
32. eway to realize the transit between two ends 3 5 2 2 Single Server Mode Endpoint Type SiPPhone Advanced Settings Config Mode Single Server Mode gt Media Settings Phone Number DisplayName sd SIP Proxy SIP Registrar Server RegisterExpingds sd Outbound Proxy p Home Domain Cr Authentication ID 120 Call Forward Type Mat Forward Call Forward Humber 1 Backup Server Enable f Disable The setting parameters relating to the SIP are as follows A Telephone Number This parameter is used to set the telephone number of the line The telephone number is a unique ID when the gateway serves as the callee B SIP Proxy Server This parameter is used to set the address of the SIP proxy server If the SIP proxy server uses the special port other than the SIP default port 5060 you can add the detailed port number at the end of the IP address or domain name of the proxy server For example 192 168 2 26 3000 or hy con com 3000 C SIP Registration Server 33 http www dbltek com GoIP GSM Series Voice Gateway The SIP registration server is a server used by the gateway to register the account This parameter is used to set the IP address or domain name of the SIP login server If the SIP registration server uses a special port other than the SIP default port 5060 you can add the detailed port number at the end of the IP address or domain name of the registration server For example 192 168 2 26 3000
33. f ms c3on the duration when the frequency 3 is on ms C3off the duration when the frequency 3 is off ms f1 the frequency of tone 1 300 to 3000Hz f2 the frequency of tone 2 300 to 3000Hz f3 the frequency of tone 3 300 to 3000Hz f4 the frequency of tone 4 300 to 3000Hz pl the increment of tone 1 0 to 31 023dB 1dB increments p2 the increment of tone 2 0 to 31 0 3dB 1dB increments p3 the increment of tone 3 0 to 31 0 3dB 1dB increments p4 the increment of tone 4 0 to 31 0 3dB 1dB increments Example To add a prompt tone where f1 is 450Hz clon is 750ms and cloff is 1000ms enter the following values in the corresponding boxes 1 0 750 1000 0 0 0 0 450 0 0 0 20 0 0 0 3 4 Network Setting Click the Network Setting in the menu on the left and the following page is displayed 29 http www dbltek com GoIP GSM Series Voice Gateway Network Configuration L AH Port Static IP ca Le optional Default Route D i pimaryons ae JE Optional 802 10 VLAN Enable C Disable VLAN ID D LL VLAN QoS e Advanced wa M T l Address Pio o Address 3 4 1 LAN Port Setting PC Port IP Address Subnet Mask DHCP Server Static IF 182 158 8 1 255 255 2551 C Enable Disable The LAN port of the GoIP can be set to the dynamic IP through DHCP fixed IP and PPPoE dialing There are three setting modes A Dynamic
34. for matching of dialing rules The rule format is A B A aa bb or A A B aa bb For example you can specify the range of numbers beginning with 2 to 8 as 2 8 aat bb or numbers beginning with 13 to 15 as 1 3 5 aa bb Examples 1 Rule 0 1 0755 a The input number is 02083185711 and the output number is 02083185711 b The input number is 83185700 and the output number is 075583185700 45 http www dbltek com GoIP GSM Series Voice Gateway 2 Rule 00 0010 0 861 86755 a The input number is 008522343318 and the output number is 8522343318 b The input number is 02083185711 and the output number is 862083185711 The input number is 83185700 and the output number is 8675583185700 le 3 Rule 00 10 0 00861 0086755 a The input number is 008522343318 and the output number is 008522343318 b The input number is 02083185711 and the output number is 00862083185711 The input number is 83185700 and the output number is 008675583185700 O 4 Rule 0 11 3 9 01 2 8 0755 0755 a The input number is 076322343318 and the output number is 076322343318 b The input number is 13044557766 and the output number is 013044557766 Or the input number is 13644557766 and the output number is 013644557766 The input number is 23185700 and t
35. he first line indicates the number to receive the SMS The second line indicates the content of the SMS MESSAGE sip 20001 192 168 2 162 5060 SIP 2 0 From sip 3999 192 168 2 89 gt tag 5031 To sip 20001 192 168 2 1 Call ID 808807EB A8B3 DD11 BBA6 005056C00008 192 168 2 89 CSeq 3 MESSAGE Contact lt sip 3999 2 192 168 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 Note The SMS forwarding mode of the GoIP is only functional under the SIP protocol 3 10 Transparent Transmission of PSTN Caller Numbers The GoIP permits the transparent transmission of PSTN caller numbers to the VoIP system in various methods CID Forward Enable Disable 64 http www dbltek com GoIP GSM Series Voice Gateway H323 Terminal nn CID Forward Mode Disable rv Disable Use Remote Party IL Use CID as SIP Caller ID SIP Terminal Disable It s not allowed to transfer the PSTN caller number to the VoIP system Enable The CID is set as the SIP caller number Use Remote Party ID The GoIP will add the PSTN caller number to the call request signaling of the VoIP system The signaling is as follows provided that the PSTN caller number is 13800000000 Sending Message to 192 168 2 1 5060 INVITE sip 5000 7192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UD
36. he output number is 075523185700 O Or the input number is 73185700 and the output number is 075573185700 3 5 7 2 Dialing Rule with Specified Length of Numbers If you need to specify the length of telephone numbers matched you can specify the dialing rule as AAXXXXXX aat tbb Where AAXXXXXX indicates the number to match and the length of the number AA indicates the head and other numbers are represented by X or x The digits after the colon indicate detailed actions to be taken on the number The setting is as follows In the above example 3 the rule is 00 10 0 00861 2 0086755 which can be changed to 00 10 0 0086l 1 8 xxxxxxx 4 0086755 It means that when you dial a number whose first digit is 1 to 8 and total length is 8 the gateway will automatically dial the number and add 0086755 before the number Examples 0 113 01 0755 46 http www dbltek com GoIP GSM Series Voice Gateway This rule enables the GoIP to add O before the mobile phone number and 0755 before the telephone number The above rule can be change to 0 113 0 9 xxxxxxxx 0l 1 8 xxxxxxx 0755 Similarly this rule enables the GoIP to add O before the mobile phone number and 0755 before the telephone number However the length of the mobile phone number is limited to 11 digits As shown above the length of telephone numbers is limited to eight digits 13 0 9 xxxxxxxx and 1 8 xxxxxxx represent
37. ing page is displayed The GoIP supports three types of outgoing call via SMS SMS Mode Dial SMS Dial Mode 1 A Mode 1 In this mode the GoIP sets the calling number of the SMS as the calling number of the call and the called number as the short message content B Mode 2 In this mode the GoIP sets the SIP number of the GoIP as the calling number of the call and the called number as the short message content 53 http www dbltek com GoIP GSM Series Voice Gateway C Mode 3 In this mode the GoIP sets the SIP number of the GoIP as the calling number of the call and the called number as the short message content and the number of the short message sender whose format is short message content the number of the short message sender D SMS Dial Prefix When the GolIP initiates the SMS call the GoIP will change the prefix number to the called number prefix Examples of SMS Dialing In the following SMS dialing examples the account of the SIP of the GoIP is set as follows Call Settings Endpoint Type SIF Phone Advanced Settings Contig Mode Single Server Made Media Settings Phone Humber 20001 ooo Display Hame SIP Proxy 192 158 2 1 SIP Registrar Server 192 166 2 1 Register Expiry s Outbound Proxy l Home Domain Authentication ID 20001 Call Forward Type Mot Forward Call Forward Humber d Backup Server C Enable Disable Mode 1 SMS Mode Dial SMS Dial
38. l Forwarding Setting on the Call Route and Authentication Mode The gateway provides the call routing function for users which can be set in the Call Forwarding Setting The call routing is to forward calls to specified numbers so that the dialing time can be decreased In addition the gateway provides three authentication modes for the uplink calls from PSTN to VoIP and downlink calls from VoIP to PSTN Do not set these parameters when they are not needed 3 7 1 Call Route Setting Forward to PSTN Enable Disable Forward to VoIP Enable Disable ree dpud O OFFENE POET VoIP To PSTN PSTN To VoIP erum 48 http www dbltek com GoIP GSM Series Voice Gateway Downlink from VoIP to PSTN Uplink from PSTN to VoIP Note The value of Call PSTN must be Enable Otherwise The GoIP prohibits any access to the PSTN Therefore set this parameter carefully The above Note is also suitable for Call VoIP 1 Set a hotline number in Call PSTN Forwarded to number When the user served by the VoIP network calls the GoIP the call is forwarded to the hotline number When the user served by the VoIP network calls the GoIP the GoIP gateway connects the call and dials 88290211 directly This feature is especially useful for hotline services 2 Set a VoIP number in Call VoIP Forwarded to number When the user served by the PSTN calls the VoIP network the call is forwarded to the VoIP number When a user calls another user
39. l with that of the admin The detailed procedure is as follows a The keyword reset and reboot is case insensitive but the password is case sensitive b When the reset instruction is sent the GoIP will automatically reboot To perform reset the admin password of GoIP user is tengda Input reset tengda or RESET tengda in the SMS to reset To perform reboot the admin password of GoIP user is tengda Input reboot tengda or REBOOT tengda in the SMS to reboot 3 Page Setting Before setting the page you need to have the IP address of the PC port of the gateway first Connect the computer for setting the gateway to the PC port of the GoIP The GoIP gateway has a built in page server that is used to accept or obtain the HTTP You can set the related functions for the GoIP through the Internet Explorer 3 1 Page Setting Menu You can access the setting page of the GoIP gateway through the IP address of the LAN port or PC port The default factory settings are as follows A The LAN port supports the DHCP dynamic IP address Users can dial the SIM card 11 http www dbltek com GoIP GSM Series Voice Gateway number of the gateway and if connected dial 00 to obtain the IP address B The default IP address and mask of the PC port are 192 168 8 1 and 255 255 255 0 respectively Enable the DHCP service of the PC port Connect the computer with the PC port of the gateway through the IP addr
40. le e O PSTN Toop 2208 VolP To PSTN ia PSTN To VoIP one as we O PSTN VoIP Forward to PSTN Password or Trust Li Forward to VoIP Auth oo word or Trust Li Auth Mode Password or Trust Li Mode sem Ld ME M VoIP To PSTN PSTH To VoIP SMS Sender PSTN Trust List SiM Card Settings PSTH Trust List YolP Trust List lt Trust Number 1 135631972345 VoIP Trust List Trust Number2 EE Trust Number 1 3306 Trust Number3 Downlink Uplink This mode is used to set the above password authentication and trust list authentication at the same time For a downlink call the authentication mode is as follows If the number in the trust list is used to dial the user served by the PSTN the call will be connected When the number you dialed is not in the trust list you need to enter the password after the secondary dial tone is played Then the call will be connected For a uplink call the authentication mode is as follows If the mobile number or fixed number in the trust list is used to dial the user served by the VoIP the call will be connected When the number you dialed is not in the trust list you need to enter the password after Please Enter the Password is played Then the call will be connected 3 8 Call Duration Limit The call duration limit is to limit the call duration of SIM cards in the gateway Through this function you can specify the total call duration of SIM cards When the call duration i
41. llow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 Mode 2 SMS Mode Dial SMS Dial Mode 2 When a short message of 8675588228822 is sent from the mobile phone number 86 13800000000 to the GoIP the GoIP will send the following call requests When the GoIP sends a call request through the SIP number of the GoIP the GoIP will automatically add the number of the short message sender to the PSTN Forwarding Number in Call Forwarding VoIP Incoming Call Forwarding to the PSTN Immediately In this mode when the GoIP receives the call from the SIP server the GoIP will forward the call to the short message sending equipment through the GSM network The SMS dial prefix is still valid in this mode The call request signaling in this mode is as follows Sending Message to 192 168 2 1 5060 INVITE sip 8675588228822 0 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 237 5060 branch2z9hG4bK 363969813 From lt sip 20001 0192 168 2 1 5060 user phone tag 65248630 To lt sip 8675588228822 192 168 2 1 gt Call ID 117025903 192 168 2 237 56 http www dbltek com GoIP GSM Series Voice Gateway CSeq 2 INVITE Contact lt sip 20001 192 168 2 237 5060 gt Max Forwards 30 User Agent DBL Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application
42. n VolP to PSTN SMS Sender Silbi Card Settings VoIP Trust Listss VoIP Trust List Trust Number Trust Number D d Trust Humber3 3 Trust Number4 a Downlink GoIP GSM Series Voice Gateway Forward to VoIP Enable Disable ame Gae O O O PSTN To VoIP gus ee VoIP Forward to MolP Auth TREE Cat m Mode PSTM Trust List PSTH Trust List Trust Humber 1 Trust Number had Trust Numbers a Trust Number4 D 1 n Trust Numbers p Trust Number6 EB 3 Trust Number E Du Uplink The setting is as follows Select Forward to PSTN Authentication Mode gt Trust List Authentication Click the VoIP Trust Number List and the VoIP Trust Number List is displayed Maximum 15 trust numbers can be entered Enter the trust VoIP number in trust number sequence If only a VoIP trust number such as 3306 is set only the number 3306 can be used to dial the PSTN from the VoIP For a uplink call from the PSTN to the VoIP the PSTN trust number shall be entered in the trust number sequence The setting is the same as that of the password authentication In this mode the Call PSTN Dialing Rule Call VoIP Dialing Rule parameters are still valid For details about the setting see section Dialing Rule 50 http www dbltek com GoIP GSM Series Voice Gateway 3 7 2 3 Password or Trust List Authentication Forward to PSTH v Enable 0 Disable Forward to VoIP v Enable Disab
43. ndicator is ON after connected with the network equipment and LAN indicator D blinks during data transmission EN This indicator is ON after connected with the network equipment and PC indicator l n blinks during data transmission 1 The RUN indicator blinks once every 100ms during startup s 2 When GoIP is connected with the server the RUN indicator blinks RUN indicator once per second http www dbltek com GoIP GSM Series Voice Gateway When the GSM module of the GoIP logs onto the local GSM network this indicator blinks once per second When this indicator blinks quickly it indicates that the GoIP is trying Channel indicator to log onto the local GSM network When the GSM channel of the GoIP is activated this indicator is normally ON 10 http www dbltek com GoIP GSM Series Voice Gateway 2 4 SMS Instructions Users can send instructions to the GoIP gateway through the SMS Function Instructions Short Remark message content Obtain INFO or info Case insensitive information from the LAN port Reset the GoIP RESET Password The keyword RESET is case insensitive a a Reboot the GOIP REBOOT Password The keyword REBOOT is case insensitive mm S 1 When info or INFO is sent to the GoIP through SMS the GoIP will return immediately the LAN port info to the mobile phone 2 When performing reset reboot through the SMS the password authentication is required The password is identica
44. noti disp c OH EE HE EN EH a DO i EE aaa 14 i WS PONS T wn ora ees aa sec orcs PE esa 15 SONNEN UU T T 15 3 9 2 MIME Zone and Dite SCLVEL ou maa em ap Ra UT pM MIDI n MI DAMM DUE 16 3 9 3 DTME Nhninium Detection Interyal os hr eo etiani ea eti eas 16 9 9 d AUEOTHIAEC OC CLIN a nates nate docto Bee a pee tol ME aset strane E ESL aed Oo Lote tede 17 2 95 Setting of tbe Remote Control uoce pn on tp tO De aut be IN Duos Pese a eds 17 3 9 0 Gap M Group MOE iur eoo oan HET eo pode PE Dip Osee T ERREUR RN Geese 18 3 3 6 Anonymity of the GSM Caller Number essere 19 SES RU ETE 19 3 39 59 MS Send tO C HODE sns nicht eu scant to oio e RA e on folet dato dol dauid 20 I AS MU BNC NP P e 21 S520 Hmn RES All oio eite ect E vieta o Po eae atte 21 25 99 bila Aiea Ode Mat HIN uitio ii toni iara ordei e ERE LUIS EIE deor Roo RES RAE RS 21 SSMUS d PEE 21 5 9 E Prompt FONE SY S06 lH ara donaret Ped E Cor mt dco oe 22 SO INGEDWOELK SOLIS dioe eH ee it eot eoe oH Ep ood EN aiit tu d NU Go dE UPS 25 SALLAN POLE SC RIO ci cera ctrceuis eie nut d REND denen ET dest e d ta D utate doeet esee 24 9 MM uuo RII D DEO TO 25 http www dbltek com GoIP GSM Series Voice Gateway SHAD Main DN M 26 2d d scconaar v DIN ato UY Cae Spa erry Coat Ta RA nL TCT e CCRT tuc abu UNDE 26 3 5 dI SEHE D costes eee onan OT S E E di tees EE E ATT T 21 204 I EE 323 PEL ANS CUEING o onset dasererat tutt
45. nternet http www dbltek com GoIP GSM Series Voice Gateway 1 2 Protocols TCP IP V4 IP V6 automatic adaptive ITU T H 323 V4 standard H 2250 V4 standard H 245 V7 standard H 235 standard MD5 HMAC SHAI ITU T G711 Alaw ULaw G 729A G 729AB G 723 1 and GSM voice coding RFC1889 real time digital transmission protocol Firewall penetration technology SIP V2 0 standard STUN Network management protocol NMP PPPoE PPP authentication protocol PAP Internet control message protocol ICMP TFTP agent protocol Hypertext transfer protocol HTTP Dynamic host configuration protocol DHCP Domain name system DNS User account authentication via MD5 Out band DTMF relay RFC 2833 and SIP INFO Mc ge m m PR Nc TA 1 05 000 5 2 1 3 Hardware Feature ARMOE high speed processor Voice coding and voice digital signal processor Two 10 100MB Ethernet ports that support the IEEE 802 3 standard and connect the LAN and PC LED that displays the status of Ethernet ports Ethernet cable SIM card that supports the GSM 900M 1800M and GSM 850M 1900M bands 1 4 Software Feature LINUX OS http www dbltek com GoIP GSM Series Voice Gateway Embedded HTTP that accesses internal parameters PPPoE dialing NAT broadband routing function DHCP client DHCP server Software online upgrade Automatic calling Supporting multiple languages Supporting outgoing SMS calls 1 5 Product Package List 1 A GoIP gateway 2 12V 2
46. oIP GSM Series Voice Gateway Click User Options and the following figure shows Preference Language s Time Zone Time Server DTMF Min Detect Time Gap Auto provision Preference Language s Time Zone Time Server DTMF Min Detect Time Gap Auto provision 3 3 1 Language English Hetwork Tones GMT 8 GSM Group Mode Anonymous IMEI C Enable Disable SMS Sender Remote Control Reboot Time User Options of the GoIP English MT 8 nal ntp arg EREE 4 C Enable Disable Remote Control Network Tones GSM CalleriD Anomymous GSh Band Reboot Time China Phone Code User Options of the GoIP_4 China Disable C Enable Disable C Enable Disable Iv VR Iv Auto Reboot 4 00 China I C Enable Disable 3001800 M Auto Reboot 4 00 C Enable Disable v WR To select a language refresh the page to enter the language page required For example the current language is simplified Chinese If you wish to display the page in English click English in the menu After your terminal is restarted all the pages will be displayed in Nn http www dbltek com GoIP GSM Series Voice Gateway English Language s English X Time Zone Time Server e AR English Click English on the upper right corner of the setting page Then the setting page will display all info in English But the shortcut will not change th
47. ology The software can be downloaded from the DBL website or obtained from our technical personnel SMS server port It is the SMS port of the SMS server The default value is 44444 It must be consistent with that of the server if modified on the server Authentication ID It is the user ID Make sure that the server has corresponding ID Authentication password It is for filling the user password Note The SMS send client of the GoIP 4 is in the call set options which needs the support of the GoIP SMS management server developed by DBL 20 http www dbltek com GoIP GSM Series Voice Gateway 3 3 7 GSM Band OSM Band 30071800 The GoIP gateway support the GSM GPRS 900 1800 and 850 1900 bands 3 3 8 Timing Restart The GoIP gateway restarts at least once at the specified time every day to clear the buffer of the GoIP so that the GoIP can operate normally Reboot Time 04 00 China Phone Code Enable Disable M YR 3 3 9 China Area Code Matching The GoIP can match all area codes of China to ensure the prompt dial The default status is Disable Reboot Time 04 00 China Phone Code C Enable Disable Iv VR 3 3 10 IVR Reboot Time 04 00 China Phone Code Enable Disable M YR By default the IVR is enabled When a call comes in the system prompts the user to dial a second time When the IVR is disabled the system will not prompt the user to dial a second time 21 http www dbltek
48. on Relay Mode The trunk agent protocol supports encryption on communications over the gateway The H323 trunk agent protocol supports encryption on signaling in different modes for details about the agent mode see section 3 5 6 3 Media NAT Traversal Note This feature needs the support of the server developed by DBL Technology 3 5 6 2 Traversal of SIP Signaling over NAT The traversal of SIP signaling over NAT firewall is classified into Signaling HAT Traversal STUNIRFC 3488 RTP Port Range A No The mechanism of firewall traversal is not supported B STUN RFC 3489 saa STUN RFC 3488 v Traversal The Simple Traversal of UDP over NAT STUN is a protocol that enables the SIP phone to detect the existence and type of the firewall installed in the computer This parameter indicates the SIP address of the STUN server C Trunk Agent The trunk agent protocol is a firewall traversal technology developed by DBL Technology It enables the products of DBL Technology to be applicable for most LANs It involves the address port user name and password 43 http www dbltek com GoIP GSM Series Voice Gateway a a NN Encryption Relay Mode The trunk agent protocol supports encryption on communications over the gateway This feature needs the support of the server developed by DBL Technology 3 5 6 3 Media NAT Traversal The media NAT firewall traversal is classified into four types
49. register status When you register through the H 323 protocol select H 323 Terminal in Terminal Type as shown in the above figure The registration mode involves the Gatekeeper and direct connection the direct connection mode is used for calls over the IP address When all lines use one number select the single server setting mode When all lines use different numbers select the line setting mode When you select the line setting mode each line can be registered to different servers The setting parameters are as follows 28 http www dbltek com GoIP GSM Series Voice Gateway A H 323 Telephone Number The value of this parameter is a decimal numeral string that is used to confirm the telephone number in the telephony network For example 191 is a valid telephone number Enter the telephone number in this parameter B Gateway Prefix When you register through the gateway prefix enter the prefix number When the prefix number is called a dialing tone is heard and then the secondary dialing is required The gateway prefix enables the one stage dialing When users dial the gateway prefix and the telephone number the gateway will automatically dial the number without the prefix For example the current gateway prefix is 123 If a user calls 075588290211 the user dials 123075588290211 on the IP phone B Display Name This parameter is used to display the name of the user who subscribes the H 323 service For example when
50. registration backup When a backup registration server exists in the user s system the user can enable this parameter When the backup server is enabled the gateway will automatically log into the backup server in case of the failure of the main Server 3 5 2 4 Trunk Gateway Mode Call Settings Endpoint Type SIP Phone Config Mode Trunk Gateway Mode v SIP Trunk Gateway SIP Trunk Gateway SIP TrunkGateway3 sd Phone Humber fs RegisterExpinis n ss AuthenticationID The trunk gateway is used to connect the VoIP network with the GSM network and convert the related protocols so that users served by the two networks can call each other A SIP Trunk Gateway1 It is the IP address of the server connected to the GoIP gateway When the registration timeout is 0 the GoIP is connected to the SIP server If the registration timeout is not 0 the GoIP logs into the SIP Trunk Gateway server through setting the telephone number authentication ID and password B SIP Trunk Gateway2 It is the IP address of the terminal connected to the GoIP gateway which can be an IP segment such as 192 168 2 X This means that all terminals connected over 192 168 2 segment can log into the GoIP and land through the direct connection between the GoIP and the GSM network C SIP Trunk Gateway3 37 http www dbltek com GoIP GSM Series Voice Gateway D E F G It is the IP address of the server connected to the GoIP gatewa
51. rt Extension This parameter is set for the special requirement of some customers If you are not sure do not set this parameter F H245 Tunnel This parameter is set for the special requirements of some customers If you are not sure do not set this parameter G Registration Mode This parameter is used to comply with different PB Xs and is not set normally Register Mode Register Multiple Mur DTMF Signaling Register Multiple Times H DTMF Signals DTMF signals are used to transmit call signals to the call switching center over the audio band The DTMF means that two different frequencies of sounds are combined to 16 types of dialing tones The telecom office or 1860 service hotline identify these dialing tones by analyzing the DSP and thus determining the dialing number There are two types of DTMFS inband DTMF and outband DTMF 31 http www dbltek com GoIP GSM Series Voice Gateway DTMF Signaling Outband Signaling QoS m Rum 1 Inband DTMF The inband DTMF transmits dialing tones and call voices together without any processing Therefore the inband DTMF transmits DTMF signals through a single way 2 Outband DTMF The outband DTMF transmits dialing tones over protocols such as RFC2833 which can ensure the validity of the transmission G Signaling QoS Quality of Service QoS is a network s capacity to provide priority services including the special bandwidth jitter control and d
52. rts the SMS forwarding through the SMS under SIP protocol After users send the short message to the GoIP through the SMS the GoIP will send the short message to the specified VoIP number automatically SMS Mode Relay Shi on E fl Humber As shown in the above figure select SMS Mode gt Forwarding and enter the VoIP number that is used to receive the SMS information The VoIP will automatically forward all the SMS information from the GSM network to this VoIP number Similarly the GoIP will automatically forward all the SMS information from the VoIP to the specified GSM mobile phone 1 The GoIP forwards the SMS from the GSM to a specified SIP number The following is an example that the GoIP forwards the SMS to the SIP 3999 The red part is the content of the SMS MESSAGE sip 3999 192 168 2 1 SIP 2 0 Via SIP 2 0 UDP 192 168 2 162 5060 branch2z9hG4bK 1967685528 From sip 20001 192 168 2 1 gt tag 667435795 To lt sip 3999 0 192 168 2 1 Call ID 2094144847 192 168 2 162 CSeq 4 MESSAGE Contact sip 20001 0192 168 2 162 5060 Max Forwards 30 User Agent DBL Content Type text plain Content Length 28 8613682626865 63 http www dbltek com GoIP GSM Series Voice Gateway 075583185700 2 The SMS sent to the GoIP from the SIP is forwarded to the specified PSTN number The following example is about the Hello world sent from SIP 3999 to 13682626800 Where in the content of the SIP message in red t
53. s longer than the specified value the calls shall not be connected to prevent the unnecessary or unsafe call charging The setting parameters are as follows 51 http www dbltek com GoIP GSM Series Voice Gateway SiM Card Settingss SIM Card Settings Lik LLL L Report Humber tig Report Time The parameters are defined as follows SIM card limit time This parameter sets the total call duration of SIM cards When the call duration is longer than the specified time by minutes the call can not be connected When this parameter is null the default call duration is infinite SIM card status reporting number The gateway can report the status of SIM cards remaining call duration through the SMS This parameter is used to specify the mobile phone number to receive the SMS SIM card status reporting time This parameter is used to specify the remaining call duration and then send the report SIM card ID This parameter is used to specify the ID of SIM cards in the short message report You can specify the mobile phone number corresponding to the SIM card or any character string as the ID One time call duration limit of SIM cards This parameter is used to specify the duration of one time calls by minutes Examples and Explanations SIM Card Settings GSM Module Information SIM Card Settings GSM1 Model SIM Card Expiry GSM1SIM NOT INSERTED SIM Card ID GSM1 Signal 24 tudinem GSMA Status LOGOUT Repor
54. sdp Content Length 226 Mode 3 SMS Mode Dial SMS Dial Mode 3 When a short message of 8675588228822 is sent from the mobile phone number 86 13800000000 to the GoIP the GoIP will send the following call requests When the GoIP sends a call request through the SIP number of the GoIP the GoIP will automatically add the number of the short message sender to the PSTN Forwarding Number in Call Forwarding VoIP Incoming Call Forwarding to the PSTN Immediately In this mode when the GoIP receives the call from the SIP server the GoIP will forward the call to the short message sending equipment through the GSM network The SMS dial prefix is still valid in this mode The call request signaling in this mode is as follows Sending Message to 192 168 2 1 5060 INVITE sip 8675588228822 8613800000000 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 237 5060 branch z9hG4bK363969813 From lt sip 20001 192 168 2 1 5060 gt user phone tag 65248630 To lt sip 8675588228822 8613902994477 192 168 2 1 gt Call ID 117025903 192 168 2 237 57 http www dbltek com GoIP GSM Series Voice Gateway CSeq 2 INVITE Contact lt sip 20001 192 168 2 237 5060 gt Max Forwards 30 User Agent DBL Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 3 9 2 SMS Dialing under the H 323 Protocol The GoIP permits
55. served by the PSTN the HT 342 calls the 3306 terminal of the VoIP When the 3306 terminal answers the call the HT 342 connects the call and the call will be connected This feature enables international roamers to answer the phone through the VoIP anywhere 3 7 2 Authentication Mode Setting The authentication mode is classified into the password authentication trust list authentication and password or trust list authentication Forward to PSTN Noauth e Auth Mode No Auth Mo Auth Password Forward to VoIP Auth 77 V Mode TEM Password Trust List Trust List Password or Trust List Son Group Mone Fassword ar Trust List Downlink VoIP to PSTN authentication mode Uplink PSTN to VoIP authentication mode 3 7 2 1 Password Authentication Forward to PSTN Password Auth Mode Password oi Fa ESTA VoIP To PSTN The setting is as follows Select Forward to PSTN Authentication Mode gt Password Authentication Enter the password in Call PSTN Authentication Password As indicated in the above figure for calls from the VoIP to the PSTN when the second dialing tone is heard dial the set password and the call will be connected For calls from the PSTN to the VoIP when Please Enter the Password is played enter the password and then the call 49 http www dbltek com will be connected 3 7 2 2Trust List Authentication Forward to PSTN Enable C Disable Dial Pla
56. t Number SIM Card State G 5M1 SIM Report Time Remain Time SIM Per Call Limit GSM1 Number SIM Card Call Duration Setting SIM Card Remaining Call Duration The setting is same as the SIM card time limit diagram The total call duration of the SIM card 52 http www dbltek com GoIP GSM Series Voice Gateway is 30 minutes When the call duration is less than or equals 10 minutes the gateway will send a short message to 13713652130 the SIM card ID is 2130 in the reporting message to report the remaining call duration When one time call duration is longer than 8 minutes the call will be disconnected When the call duration is over it becomes 0 Users can dial the SIM card number by the mobile phone and when the second dialing tone is heard press 10 to restore the value 3 9 SMS Mode The GoIP permits you to call VoIP users or forward short messages through the SMS 3 9 1 SMS Dialing under SIP Protocol Under the SIP protocol the GoIP permits users to dial back through the SMS After users send the called number to the GoIP through the SMS the GoIP gateway will send a call request to the SIP server automatically Users who need this function should choose the following parameters Forward to PSTN Enable Disable SMS Mode Disable uni meer eObpronwana VoIP To PSTN eee 7 7 RE PSTH Forward to PSTN No Aut Auth Mode PORUM SIM Card Expiry Select SMS Mode gt Dial and the follow
57. t is used to specify the size of the media packet The unit is ms the actual number of bytes depends on the compression algorithm C Jitter Delay Processing Mode This parameter is used to specify the algorithm model of the jitter delay buffer The Adaptive mode should be set Other modes are only used for tests and should not be set in actual applications Jitter Buffer Fixed Delay ms Adaptive Media QoS Sequential D Media QoS Quality of Service QoS is a network s capacity to provide priority services including the special bandwidth jitter control and delay used for real time and interactive traffic and improvement of the packet loss ratio This parameter is used to mark the specified QoS label for the voice packet to increase the network service quality Media QoS Media Encryption Note For details about media encryption and media NAT penetration refer to 3 5 6 Firewall penetration 3 5 5 Voice Coding and Sequence This parameter is used to modify the compression coding according to the requirements of the service provider Audio Codec Preferences M alaw M ulaw I grea M gF2da Iv gz22ab Iv 97231 If a compression coding is ticked it indicates that the compression coding is available The UP and DOWN are used to adjust the priority of the selected voice compression coding 4 http www dbltek com GoIP GSM Series Voice Gateway 3 5 6 Firewall Penetration In the
58. the service provider If the PPPoE is set the main DNS will be automatically provided by the service provider This parameter can be null 3 4 4 Secondary DNS When the main DNS address fails to connect or is not available the secondary DNS can be used such as 202 67 156 222 or obtain from the service provider If the PPPoE is set the secondary DNS will be automatically provided by the service provider This parameter can be 26 http www dbltek com GoIP GSM Series Voice Gateway null 3 5 Call Setting This section describes the basic setting of the network connection relating to the GoIP gateway which supports two protocols H323 and SIP The setting page is as follows You can select a protocol in the Terminal Type Endpoint Type SIP Phone Contig Mode 3 5 1 H 323 Terminal Setting The H 323 protocol involves the direct connection mode and Gatekeeper mode 3 5 1 1 Direct Connection Endpoint Type H323Phone Advanced Settings gt Endpoint Mode DiectMode v Media Settings Phone Humber pisplayName id um Under this mode the GoIP operates in the point to point status The setting parameters are as follows A H 323 Telephone Number The value of this parameter is a decimal numeral string that is used to confirm the telephone number in the telephony network For example 5551234 is a valid telephone number Enter the telephone number in this parameter B Display Name
59. tions dpe Content Length 226 REGISTER 3 11 Save the change After setting is changed click Save and the new setting will be valid Otherwise the new setting is invalid Preference Status Language s Heti bi a I Configurations Time Zone GMT 8 Time Server panl ntp org i a DTE Min Detect Network Hmesen Auto provision C Enable Disable Call settings Remote Control I Call Divert cave Changes Discard Changes Tools Network Configuration LAN Port DHCP l 802 14 VLAN C Enable Disable Anvanran Note Some of parameters of the gateway will not be valid until the gateway is restarted Therefore you are advised to restart the gateway after the parameters are modified so that the modification can take effect 3 12 Abandon the change When the new setting is not saved you can clear all the unsaved parameters 66 http www dbltek com GoIP GSM Series Voice Gateway 3 13 Tool Select Menu gt Tool The following page is displayed sis Configurations Last Upgrade Time Current Version GHS 4 01 12 Tools Online Upgrade Change Password Reset Config Reboot 3 13 1 Online Upgrading Warning Only experienced users and administrators can implement the online upgrading Select Tool Online Upgrading The online upgrading page is displayed as shown in the following figure Enter the complete name and path of the upgrade package such as http
60. users to dial back through the SMS under the H 323 protocol After users send the called number to the GoIP through the SMS the GoIP gateway will send a call request to the H 323 GK automatically Users who need this function shall choose the following parameters SMS Mode Dial M SMS Dial Mode 1 Select SMS Mode Dial and the following page 1s displayed The GoIP supports three types of outgoing call via SMS A Mode 1 The current version of the H 323 protocol does not support this mode but the later version will In this mode the GoIP sets the number of the short message sender as the calling number of the call and the called number as the short message content B Mode 2 In this mode the GoIP sets the H 323 number of the GoIP as the calling number of the call and the called number as the short message content C Mode 3 The current version of the H 323 protocol does not support this mode but the later version will 58 http www dbltek com GoIP GSM Series Voice Gateway In this mode the GoIP sets the H 323 number of the GoIP as the calling number of the call and the called number as the short message content and the number of the short message sender whose format is short message content the number of the short message sender D SMS Dial Prefix When the GolIP initiates the SMS call the GoIP will change the prefix number to the called number prefix Examples of SMS Dialing In the following
61. ver which needs the support of the specific system For details please contact the 17 http www dbltek com GoIP GSM Series Voice Gateway technical support of DBL Technology 3 3 6 GSM Group Mode Users can establish a GSM group containing multiple GoIP gateways Under this mode the administrator only needs to provide a GSM number to the user to call in the VoIP system Client Mode GolP Client Mode GolP Client Mode GolP I1 1T 1 Incoming call forward to Client GolP Each GoIP can operate in any of the following modes GSM Group Mode Disable GSM CallerID Anonymous IMEI Prohibit This mode is used when the GoIP operates independently Serve as the server When GoIP operates in this mode the administrator only needs to provide the user with a GSM number of the GoIP as a unique access number to the GoIP group In one GoIP group only one GoIP gateway can be used as the server When the GoIP serves as the server the GSM unconditional call 18 http www dbltek com GoIP GSM Series Voice Gateway forwarding or busy call forwarding can be activated The unconditional call forwarding is used to forward all incoming calls to other clients of the group The busy call forwarding is used to forward incoming calls to other clients of the group when the status of the SIM card of the server is ACTIVE GSM Group Mode As Server GSM Forward Mode I Incanditianal Farwar GSM CallerID Unconditional Forward
62. w Pass Confirm Pass Tools Online Upgrade Change Password Administrati Hew Pass Reset Config Confirm Pass 68 http www dbltek com GoIP GSM Series Voice Gateway Select Tool gt Reset to restart the GoIP Microsoft Internet Explorer X v Are vou sure to reboot Ehe device 4 Parameters of Equipment LN sf Default 900M IROOM Defaul GSM band Optional 850M 1900M Customized Power mp LED RUN GSM LAN PC RUN GSM LAN PC Network adapter LAN 100 10BASE T NT ask Without DC Adapic Operating 0 40 C 0 40 C temperature Operating 4096 9090 Not Congealed 4096 9096 Not Congealed 2e 1 69 http www dbltek com GoIP GSM Series Voice Gateway 5 Factory Default Parameter Table LAN Dynamic IP DHCP Network Fixed IP 192 168 8 1 Password 70
63. y Telephone Number This parameter is used to set the telephone number of the line The telephone number is an unique ID when the gateway serves as the callee and takes effect when the GoIP logs into the SIP Trunk Gatewayl Registration Timeout s When the registration timeout is 0 you cannot register the gateway and the gateway will be connected to the server directly You can reference the setting parameters of the single server mode to register the gateway Authentication ID This parameter is used to set the authentication account when the gateway logs into the SIP Trunk Gateway proxy server The parameter can be null in the case of interconnection Password This parameter is used to set the authentication password when the gateway logs into the SIP Trunk Gateway proxy server The parameter can be null in the case of interconnection 3 5 3 SIP Advance Setting The advance setting of the SIP involves the signaling and media Users can set according to their special requirements Select SIP Menu Advance Setting Media Advanced Settings lt lt Local Signaling Port 5060 i sip 183 NAT Keep alive Enable Disable Reigster Mode Mode 1 Advanced Timing DTMF Signaling inand o Signaling QoS None o Signaling Encryption Noe am us s Media Settings 38 http www dbltek com GoIP GSM Series Voice Gateway A Signaling Port SIP Local Port The SIP local port is the local UDP
64. y also be caused 3 3 4 Automatic Setting If the service provider provides the automatic setting you can select Enable to start the automatic setting feature and enter the address of the server If the service provider does not provide the automatic setting you need to select Disable to speed up the startup time of the GoIP Auto provision Enable Disable provision Server E i Provision interval It is a special server which needs the support of the specific system 3 3 5 Setting of the Remote Control Press 20 on the terminal to initiate the request to realize the remote management of equipment The remote control server is provided by the service provider The default port is 1920 and the terminal is identified by the SN The remote control password is identical with that of the server and is set as default Remote Control lt lt Remote Sever Remote Server Port D 0 E Remote Server ID ee Remote Server Key fo In the following figure the remote control server is set as 202 155 200 154 The terminal user presses 20 and a long tone is heard which indicates that the instruction has been successfully sent The remote administrator access http 202 155 200 154 8086 and the model and SN of the gateway are displayed Click the gateway SN to set the remote gateway Remote Cantrole Remote Server Remote Server Port Remote Server ID Remote Server Key Note It is a special ser

GoIP Series SIM Card for GSM Voice Gateway User Manual

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