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DBL Technology GoIP-1 User Manual Download Page 67

DBL Technology Co., Ltd. 

 

  GoIP User Manual 

www.dbltek.com 

                                                            Technical Support: [email protected] 

 

67 

Appendix D.

 

GSM Group Mode 

The GSM Group mode is designed to simulate the function of one GSM number with multiple lines.    The idea 
is to form a GSM group with one number being the "Server".    Only this GSM number is announced to the 
public.    Calls  to  this  number  are  forwarded  to  other  GSM  numbers  ("Clients")  in  the  group  until  all  GSM 
channels  are  used  up.    Effectively  speaking,  if  there  are  40  GSM  channels  in  a  group,  a  maximum  of  40 
concurrent calls can be achieved by just calling the GSM number of the Server channel.    The diagram below 
demonstrates this concept with only single channel GoIPs.    In fact, GoIP with multiple channels can also be 
used.    Only one "Server" in a group and all the other channels must be set to "Client" individually. 

 

 
 

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Summary of Contents for GoIP-1

Page 1: ...ual www dbltek com Technical Support support dbltek com 1 GoIP User Manual VoIP GSM Gateways Models GoIP GoIP 4 4i GoIP 8 8i WoIP 8 GoIP 16 GoIP 32 Revision 1 5 2016 6 24 https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 2: ...3 Configuration 18 5 3 1 Preference 18 5 3 2 Network 21 5 3 3 Basic VoIP 23 5 3 4 Advanced VoIP 29 5 3 5 Media 32 5 3 6 Call OUT 34 5 3 7 Call OUT Auth 37 5 3 8 Call IN 38 5 3 9 Call IN Auth 41 5 3 10 SIM 42 5 3 11 SIM Forward 44 5 3 12 IMEI 45 5 3 13 SMS 46 5 3 14 GSM Carrier 49 5 3 15 GSM Base Station 49 5 3 16 Event Trigger 51 5 4 Tools 53 5 4 1 Online Upgrade 53 5 4 2 Change Password 53 5 4 3 ...

Page 3: ...ckup Restore 59 5 4 12 Reset 60 5 4 13 Reboot 60 Appendix A Special SMS Commands 61 Appendix B SMS To VoIP 62 Appendix C Custom Network Tones 66 Appendix D GSM Group Mode 67 Appendix E CID Call Forward 68 Appendix F Volume Adjustment 69 https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 4: ...ignal reception which causing poor performance Please note that the antenna for WoIP 8 is about 4 inches long and the one for GoIP 8 is just 2 inches long The device configurations for WoIP is in general the same as those in GoIPs 3 General 3 1 Introduction GoIP and WoIP are abbreviated from GSM over IP and WCDMA over IP respectively They are new types of VoIP gateway that allows call terminations...

Page 5: ...ly insert your SIM card to a GoIP WoIP You can then setup the GoIP WoIP to forward all incoming calls to another GSM number in the world via a VoIP service provider The charge per call from a VoIP service provider is significantly lower than the roaming charge For office environment GoIP WoIP offers a quick way to replace the traditional PSTN lines or T1 E1 lines to your IP PBX There is no initial...

Page 6: ...Built in SIP Proxy Simplified PPPoE Dial Up Router function DHCP client Server QoS VLAN VPN PPTP Online firmware upgrade Remote Control Mechanism for remote technical support Proprietary Auto Provisioning Mechanism Remote SIM function Short Messages SMS support standalone and server based Call Management and Routing 3 5 Package Content Use care when unpacking the device package in order to avoid d...

Page 7: ...support dbltek com 7 Item Appearance Description 1 GoIP 1 Channel GoIP 4 4 Channel GoIP 8 8 Channel WoIP 8 8 Channel GoIP 16 16 Channel GoIP 32 32 Channel 1 x Main Unit 2 AC DC Power Adapter GoIP1 12V 500mA https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 8: ...t LED Label Description Power This LED is red and illuminates when power is connected LAN This LED is red and illuminates when the LAN port is connected and blinks when data transmission occurs PC This LED is red and illuminates when the PC port is connected and blinks when data transmission occurs RUN This LED is green and blinks at a rate of every 100ms when VoIP is not ready for making calls Fa...

Page 9: ...ch GSM channel individually via the web interface 1 SIM card slots are located either at the bottom for old hardware or at the back for new hardware of the main unit For the models with the SIM card slots located at the bottom you need to open the bottom SIM cover in order to install SIM cards First slide the metal clip to the direction as indicated on the top of the clip Insert a SIM card to each...

Page 10: ...shown in the figure below The cut corner is pointing downward with the metal contacts facing the front of the GoIP For GoIP 32 the SIM card insertion orientation is shown in the figure below The cut corner is pointing downward with the metal contacts facing the front of the GoIP https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 11: ...s the LAN port In Router mode the PC port is set to a different network segment In this case please make sure that the PC network segment IP 192 168 x is different from the one in the LAN port network 4 The DC port is for power connection Please only use the AC DC adapter provided Adapter with different rating or vendor may damage the device or affect its performance 5 The Reset button is recessed...

Page 12: ...le Once the call is answered dial 01 to hear a voice prompt reporting the LAN port IP address ii Send the INFO SMS command to one of the GSM channels available An SMS with the LAN port IP address is sent back to the message sender Please refer to the Appendix A Special SMS Commands for more information Once the LAN IP address is known you are now ready to access its built in http web server by typ...

Page 13: ...e the login password to a more complicated pattern 4 Current Time This shows the current time which is obtained from the network time server specified If the time is not correct there may be an issue with the network connection There are four pages under the Status section 1 Summary 2 General 3 GSM 4 SIM Call Forward All 4 status pages are updated automatically in every 5 seconds It is important t...

Page 14: ...econd dial tone prompts the caller to press a phone number The Status changes to ACTIVE since the start of the second dial tone till a phone number is received for dialing or the call is terminated d DIALING phone number This occurs when the GoIP is dialing out a phone number via the corresponding GSM channel or The DIALING status shows that a number is being dialed out via the corresponding GSM c...

Page 15: ...Model This field shows the model number of the device d Local Time This shows the current system time It is a good indication for normal network access provided that the network server address and time zone are set properly Please make sure that these information are provided when reporting a problem or requesting for technical support 2 Network a LAN Port This field shows the IP address assigned ...

Page 16: ...server d Routing Prefix This shows the current setting for the Routing Prefix If it is set the Routing Prefix is used to select the corresponding channel to dial out a call Please refer to Section 3 3 3 for more information GSM e CH This corresponds to the physical GSM channel number f Login This shows the current GSM Registration status for voice calls g Call In This shows the Call IN setting for...

Page 17: ... For more information please refers to the section 3 3 15 9 Cell ID This shows the Base Transceiver Station BTS ID 10 LAC This shows the Location Area Code The bottom table shows more detailed information on the onboard GSM modules and the SIM card inserted 1 Module The model number of the GSM module 2 Firmware Ver The version number of the firmware installed in the module 3 SIM Number The GSM num...

Page 18: ...t support dbltek com 18 5 3 Configuration Click Configuration on the left hand column to display the Configuration page Under this section there are 16 items in the submenu as shown below 5 3 1 Preference https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 19: ... 5 Remote Control Remote Server Remote Server Port Remote Server ID Remote Server Password This is a unique feature that allows remote access to the device s built in Web server even when it is installed behind NAT To achieve this function a Remote Control Server is required to be installed This server is a free Linux based utility and is available for download via our website Please contact techn...

Page 20: ...support if you need further help on this The default DDNS Address is voipddns net which a free service offered by the manufacturer Please contact technical support if you want to install your own DDNS server The default communication port number is 39800 This specifies the interval between registrations to the DDNS voipddns net 39800 120 mins 9 Auto Reboot Reboot Time This option allows the device...

Page 21: ...ork conditions short dropouts may occur due to packet jitter loss Therefore DTMF digit may be detected more than once if these dropouts are not taken into account Consequently the call is dialed to an incorrect number To avoid this problem a dropout window is used to avoid false detection when dropouts occur During this window the same DTMF digit is not recognized more than once The range of the d...

Page 22: ... offers both Router and Bridge modes to meet your requirements 1 Static IP default setting This mode enables the device to create another network segment and it then functions as a router gateway for this new network segment Select Static IP for this new segment and then enter the PC port IP address and Subnet Mask accordingly It also has a built in DHCP server to assign IPs to the devices attache...

Page 23: ...atest firmware upgrade links available In general it is important to understand your VoIP application with the device before proceeding to device configuration If the device is going to work with a IP PBX please make sure that you know how to configure your IP PBX It is very important that you send us your application requirements in full details when seeking for technical support in configuring t...

Page 24: ...is found the call is rejected by sending back a SIP 503 message b Only some Routing Prefixes are set Try to match the number received for making an outgoing call against those Routing Prefixes that are set If only one match is found the corresponding channel is used to make the outgoing call If more than one matches are found the best available channel among the matched channels is selected If no ...

Page 25: ...IP Registrar Server address This specifies the backup Home Domain address Disabled 12 Routing Prefix This parameter is used for call routing When this is set the corresponding channel is only used to dial out a phone number with the matching Routing Prefix Syntax Prefix1 Prefix2 Prefix3 where Prefix is a text string which consists of digits alphabets and special characters The maximum length of th...

Page 26: ... all models except GoIP 1 This mode is basically a combination of Single Server mode and Config By Line mode It allows lines to be split up into groups Each group only uses one SIP registration for all the lines assigned to the group Each line can be assigned to only one of the 4 predefined groups in the Grouping section In this mode the Routing Prefix is assigned to the Group rather than to the C...

Page 27: ...y not work properly depending on the SIP Server and the router configurations on each side as well In this case the SIP server is required to support NAT Both routers should also be setup to map the signaling port and media ports to the SIP Server and the GoIP properly Otherwise VoIP calls may fail to establish properly in this network environment The device accepts calls from up to 3 IP addresses...

Page 28: ... in this mode are listed in the table below Parameter Trunk Gateway mode Description Default Value 1 SIP Trunk Gateway1 This specifies the first SIP Trunk Gateway address Use X or x for the last part of the address to specify the entire segment 0 255 2 SIP Trunk Gateway2 This specifies the second SIP Trunk Gateway address Use X or x for the last part of the address to specify the entire segment 0 ...

Page 29: ...Mode is set to Fixed 5060 2 SIP INVIITE Response One of the key function of the device is to allow call terminations from VoIP to GSM In general a VoIP caller dials a PSTN or GSM number E 164 number and the SIP Server routes this call to the device by sending a SIP INVITE message This parameter specifies the response to the INVITE message The three possible responses are described in details below...

Page 30: ...he device Choose Enable to activate this SIP proxy to accept any SIP registrations with the correct password which is specified in the parameter Password There is no need to create a SIP account in this server Users will have to manage the SIP numbers used on their own This facilitates the setup of a simple SIP network for customers who do not have their own SIP servers This sets the password for ...

Page 31: ... This timer specifies the timeout for an unanswered call A SIP 408 Request Timeout command is sent to the SIP Server when this timer expires Note The default value is set the maximum value so that it will not interfere with the call unanswered timeout at SIP Proxy or PBX NICT Non Invite Client Transaction RFC 3261 Section 17 1 2 ICT Invite Client Transaction RFC 3261 Section 17 1 1 Round Trip Time...

Page 32: ...ction allows the user to program various settings for media voice transmission and format Depending on your network environment and condition you may or may not need to change these settings Please see the parameter table below for more information https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 33: ... mode the jitter buffer function looks at the packet timestamp for dropped or out of sequence packet problems The data packets are sorted based on the packet timestamp 3 Adaptive The adaptive mode optimizes the size of the jitter buffer delay and depth in response to network conditions in addition to the sequential mode functions This specifies the fixed jitter delay for both Fixed and Sequential ...

Page 34: ...preference shown below a law law G 729 G 729A G 729AB G 723 1 5 3 6 Call OUT The Call Out page defines how each GSM channel handles calls when they are routed from VoIP This section MUST Be defined properly in order to enable each GSM channel to dial out calls based on your requirements In general you can achieve the followings 1 Forward all incoming VoIP calls to a fixed GSM or PSTN number 2 Dial...

Page 35: ...f an INVITE message Please refer to Section 3 3 3 1 for more information Calling the SIP number directly routes the call to the corresponding line immediately If this parameter is blank and the Callee Number equals to the SIP Number defined for this line a second dial tone is generated to wait the caller to dial a phone number Please note that this could only happen in the Single Server mode Confi...

Page 36: ... the prefix for matching starting from the beginning of the Callee Number The right portion b c is the action to be taken Both b and c are optional b means that b is removed from the beginning of the Callee Number If b is not found starting from the beginning of the Callee Number c means that c is added to the beginning of the Number generated from the last action In order for this rule to be mean...

Page 37: ...dless whether the call is completed successfully or not 2 Answered Calls The Sleep Interval is activated whenever an outgoing call is answered 5 3 7 Call OUT Auth The Call Out page defines how each GSM channel handles calls when they are routed from VoIP This section MUST Be defined properly in order to enable each GSM channel to dial out calls based on your requirements In general you can achieve...

Page 38: ...e prompt to wait for further inputs from the caller The GSM call is answered and GSM charges may apply This is referred as the Second Dial mode 4 Forward incoming GSM calls to the host channel to other idle GSM channels This enables a multi channel GoIP and or multiple GoIPs to simulate the Calll Center function Therefore only the phone number of the host channel is released for the public to call...

Page 39: ...arding to VoIP Number This parameter enables or disable an incoming GSM call to the selected channel to be forwarded to the SIP extension specified If this parameter is blank incoming GSM calls are not forwarded automatically Instead they are answered with a dial tone or a voice prompt IVR parameter enabled in the Preference page the device answers an incoming GSM call This operation is called Sec...

Page 40: ...ode Forward Mode Backup Host Address Hunt Group operation is discussed in details in Appendix D Please note that Hunt Group Mode is a property of each GSM channel and is required to be set individually Host This enables the channel selected to be the Host of the Hunt Group operation All clients registers and update the host on their channel status The host then maintains a list of clients status a...

Page 41: ...r Description Default Value 1 Call In Auth This parameter defines how incoming GSM calls are authenticated before routing them to the VoIP network connected Five options are available 1 None No authentication is required all incoming GSM calls are routed to VoIP 2 Password The caller is prompted for entering the password before the call is routed or a second dial tone is generated 3 Whitelist Only...

Page 42: ... automatically duplicates the same settings to all other channels with the exception that the SMS Alert ID is incremented by 1 with respect to the ID of the previous channel The parameters for this section are described in details in the table below Parameter SIM Card Description Default Value 1 GPRS Registration This enables the GPRS registration mode for all GSM channels Enable 2 SIP Registratio...

Page 43: ...ample 2 08 00 12 00 100 12 00 20 00 200 In this example the first period starts from 08 00 till 12 00 with a talk time limit of 100 minutes The second period starts from 12 00 till 20 00 with a talk time limit of 200 minutes From 20 00 till 08 00 the next day there is no talk time limit 3 Talk time allowed monthly This sets the monthly accumulative talk time allowed monthly The accumulative talk t...

Page 44: ...d all incoming calls when the SIM cannot register to the network There are 3 choices for each Call Forward condition 1 Set This enable the Call Forward function This setting is sent to the network whenever the SIM is starting a new registration to the network 2 Disable This disable the Call Forward function This setting is sent to the network whenever the SIM is starting a new registration to the ...

Page 45: ...the manufacturer and the brand model The following six digits CCCCCC are the serial number The last digit is the check digit which is calculated according to Luhn formula The Check Digit is calculated of all other digits in the IMEI The purpose of the Check Digit is to help guard against the possibility of incorrect entries to the CEIR Central Equipment Identity Register and EIR equipment The chec...

Page 46: ...xpires the first time after the last is pressed The default value for Change Period is 60 i e 1 hour The minimum value allowed is 10 A random IMEI only the portion CCCCCC is changed TAC portion remain unchanged can also be assigned by enabling the option Set Random IMEI when Module Powers up This feature is important to make sure that a new IMEI is assigned to each GSM channel whenever it powers u...

Page 47: ...tion This parameter specify which SMS Dial mode is used Please refer to Appendix B for more information on the modes available The parameter is applicable for SMS Dial mode It allows a prefix to be added to the phone number of the called party 2 Forward Function This mode forwards incoming GSM SMS to a SIP terminal and a GSM number Incoming GSM messages SMS received are forwarded to the SIP Phone ...

Page 48: ... be created in the SMS Server This specifies the login password for the SMS Client ID 44444 2 SMS Number Plan The function of this parameter is similar to the Dial Plan for calling It is used to modify the phone numbers of the SMS recipients It has the same syntax as the Dial Plan which is described in the Section 5 3 6 for more information When no rules are matched SMS will be sent to the origina...

Page 49: ...provider selection mode The factory default setting is Auto for automatic selection of GSM service provider based on the default preference set by the SIM card When a GoIP is installed at a location that is close to a country border it is possible that the default service provider is not selected based on the base station signal strength The GoIP may then register to a GSM service provider that ch...

Page 50: ...maximum number of channels in the Polling Channel List 4 2 Channel Switching Interval m This defines the duration in minutes when the next base station switching occurs Other than a single value a range can also be specified using the format min max where min is the minimum interval and max is the maximum interval A random value in the range specified is then assigned when a channel switching occu...

Page 51: ...e will perform the corresponding action in the same channel The Event Triggers defined are shown below Defined Events 1 Channel ACD is less than the time in seconds specified 2 Channel ASR is less than the percentage specified 3 This specifies a time duration to repeatedly execute the action specified 4 This refers to outgoing GSM calls that has no ringback tones returned from the network and the ...

Page 52: ...and then the number consecutive calls are allowed When the preset conditions are met the predefined action is executed Defined Actions 1 Null No action 2 Disable VoIP connection This causes the GoIP to deregister from the SIP Server for the corresponding line 3 Disable Call Out via GSM This prevents calls from dialing out from the corresponding GSM channel 4 Shut down GSM Module This shuts down th...

Page 53: ...tomatically Please wait patiently as this process may take a few minutes Note It is important NOT to disconnect the power during a firmware upgrade since the internal Flash may be corrupted If this happens pleases contact technical support for assistance Please reboot the device if an upgrade attempt fails before performing another upgrade 5 4 2 Change Password Click Change Password to change the ...

Page 54: ...e SIM GSM number are displayed The last option All Lines means that all channels are selected and the same USSD command is sent via all channels provided that they are in the LOGIN Status b Enter the USSD command c Click Send Example For the service provider PCCW in Hong Kong the USSD command to check balance is 122 Enter 122 and the click Start The following screen is then displayed A few seconds...

Page 55: ...us and the SIM GSM number are displayed The last option All Lines means that all channels are selected and the same SMS is sent via all channels provided that they are in the Login Status b Enter the recipient s phone number GSM c Type the SMS message in the SMS Content box The maximum length of a message is 140 characters for 7 8 bit ASCII code and 70 characters for 16 bit Unicode d Click Send to...

Page 56: ... to the SIM card 2 Disabling a GSM channel temporary Click GSM Channel Shut Down to access the webpage below to shut each GSM module individually Place a check mark to select the desired channel and then click Save to activate the shut down Remove the check mark and then click Save to turn on the channel again The All Channels selection is a short cut to turn on or to shut down all channels 5 4 8 ...

Page 57: ...nter the call duration in the Duration field When the call duration of an outgoing reaches the time specified it will be terminated automatically 5 4 10 Get Number Get Number enables the GoIP to retrieve the phone number of the SIM card connected Both SMS and USSD methods are supported however the actual procedures in getting the phone number may still vary between carriers Please find out the act...

Page 58: ... phone number 5 Carrier sends back an SMS response with the phone number However the CID of the SMS Response must be set so that the GoIP can check the SMS from this CID 6 Set the Text before the SIM phone number so that the GoIP can extract the phone number from the SMS 7 Press to save the configuration If your Carrier does not support SIM card phone number retrieval you can still obtain the SIM ...

Page 59: ...e Number the phone number of another GoIP channel 5 Set SMS Content Trigger String for an SMS Reply 6 Set CID of the SMS Response the phone number of another GoIP Channel same as the one set in SMS Service Number 7 Set Text before the SIM Phone Number Reply Content 5 4 11 Backup Restore The device configuration can be backup or restore via this page Click Backup Resotre Confguration to access the ...

Page 60: ...ry default Click OK in the pop up window shown below to confirm this action Click OK to reset the device configuration back to the factory default 5 4 13 Reboot Click Reboot to restart the device Click OK in the pop up window shown below to confirm this action The reboot process will take couple of mins https arttel ru https arttel ru shop gsm shlyuzy gsm shlyuz goip 4 ...

Page 61: ...rizes the SMS command syntax and are not part of command text SMS Message Content Function INFO Sends an SMS response to the sender with the LAN port IP address info RESET password Reset the device configuration back to the factory defaults and then reboot the device password is the password for the administration level reset password REBOOT password Reboot the device password is the password for ...

Page 62: ...ll conversation Example SMS content 8675588228822 Sender s number 861380000000 SIP Server IP 192 168 2 1 SIP Number 20001 The INVITE message sent to the SIP Server is 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 user phone tag 65248630 To sip 8675588228822 192 168 2 1 Call ID 11702...

Page 63: ...ontent Type application sdp Content Length 226 c Mode 3 SIP Message format The To field in the SIP INVITE message contains the phone numbers of both the called and calling parties These two numbers are concatenated by using the asterisk character with the number of the called party in the front The From field contains the SIP number of the line that is associated with the GSM channel Example SMS c...

Page 64: ...ip 20001 192 168 2 1 tag 667435795 To sip 3999 192 168 2 1 Call ID 2094144847 192 168 2 162 CSeq 4 MESSAGE Contact sip 20001 192 168 2 162 5060 Max Forwards 30 User Agent DBL Content Type text plain Content Length 28 8613682626865 075583185700 Please note that the SIP Server side must be programmed to process this SIP MESSAGE according to the application needed It can forward the message to the SI...

Page 65: ...20001 192 168 2 162 5060 SIP 2 0 From sip 3999 192 168 2 89 tag 5031 To sip 20001 192 168 2 1 Call ID 808807EB A8B3 DD11 BBA6 005056C00008 192 168 2 89 CSeq 3 MESSAGE Contact sip 3999 192 168 2 89 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 https arttel ru https arttel ru s...

Page 66: ...d based on the on off pattern defined 0 means infinite p1on is the tone on duration for the first frequency tone ms p1off is the tone off duration for the first frequency tone ms p2on is the tone on duration for the second frequency tone ms p2off is the tone off duration for the second frequency tone ms p3on is the tone on duration for the third frequency tone ms p3off is the tone off duration for...

Page 67: ...umber are forwarded to other GSM numbers Clients in the group until all GSM channels are used up Effectively speaking if there are 40 GSM channels in a group a maximum of 40 concurrent calls can be achieved by just calling the GSM number of the Server channel The diagram below demonstrates this concept with only single channel GoIPs In fact GoIP with multiple channels can also be used Only one Ser...

Page 68: ...IP INVITE message Choose this if both SIP Server and SIP terminal support this parameter Example Caller ID number 13800000000 The Remote Party ID parameter is included in the SIP INVITE Message below 2 USD CID as SIP Caller number This parameter specifies the use of GSM Caller ID instead of its SIP number in the INVITE message when making a call Please make sure that the SIP server supports this t...

Page 69: ... controlled by the input gain and the output gain respectively An increase in the output gain means that the GSM PSTN party hears a higher audio level An increase in the input gain means that the VoIP party hears a higher audio level Please note that changing these gain settings affects the DTMF tones in the corresponding path as well As a result DTMF tones for phone dialing may not be detected co...

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