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GoIP User Manual

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a) All Routing Prefixes are set

- Try to match the number received for making an outgoing call against the Routing Prefix of each

channel.

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 match 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 match is found, an idle channel without a Routing Prefix

is used to make the call. Otherwise, the call is rejected by sending back a SIP 503 message.

c) None of the Routing Prefixes are set

- An idle channel is selected to dial out the call.

If no idle channel is available, the call is rejected

by sending back a SIP 503 message.

It is important to note that the Routing Prefix (P) must be removed via the dial plan before the number is
dialed out.

The dial plan syntax to remove the Routing Prefix is "P:-P|".

Please refer to section 3.3.6 for

more information on the Call Out Dial Plan.

Parameter
(Single Server Mode)

Description

Default Value

1. Phone Number

This is the SIP number used by the device.

2. Authentication ID

The name of the device used in Caller Identification is defined here.

3. Display Name

The Authentication ID used for SIP registration is specified here.

4. Password

The password used for SIP registration is specified here.

5. SIP Proxy

The domain name or IP of the SIP Proxy or Server is specified here.

If the SIP Proxy is

using the standard 5060 signaling port, then there is no need to specify the port

number. Otherwise, the port number can be specified by adding ":" and then the port

number at the end of the SIP Proxy address.

6. SIP Registrar

The address of the SIP Registrar Server is specified here.

Summary of Contents for GoIP

Page 1: ...GoIP User Manual http www dbltek com Sales sales dbltek com Support support dbltek com GoIP User Manual VoIP GSM Gateways Models GoIP GoIP 4 4I GoIP 8 8I GoIP 16 GoIP 32 Revision 1 4B 2014 3 04 ...

Page 2: ...SIM Call Forward 14 3 3 Configuration 15 3 3 1 Preference 15 3 3 2 Network 18 3 3 3 Basic VoIP 19 3 3 4 Advanced VoIP 24 3 3 5 Media 29 3 3 6 Call OUT 32 3 3 7 Call OUT Auth 34 3 3 8 Call IN 35 3 3 9 Call IN Auth 38 3 3 10 SIM 39 3 3 11 SIM Forward 41 3 3 12 IMEI 42 3 3 13 SMS 43 3 3 14 GSM Carrier 45 3 3 15 GSM Base Station 45 3 4 Tools 47 3 4 1 Online Upgrade 47 3 4 2 Change Password 48 3 4 3 Se...

Page 3: ...p www dbltek com 2 3 4 9 Reboot 52 Appendix A Special SMS Commands 53 Appendix B SMS To VoIP 54 Appendix C Custom Network Tones 58 Appendix D GSM Group Mode 59 Appendix E CID Call Forward 60 Appendix F Volume Adjustment 61 ...

Page 4: ... On the other hand you can also make a call from the local telephone network to a GoIP the GSM phone number and then dial another number via a VoIP network to anywhere in the world In these two cases a VoIP Service provider is required for one side of the call termination For two fixed locations it is possible to setup GoIPs at both ends for call terminations without subscribing to a VoIP Service ...

Page 5: ...ame System DNS User Account Authentication via MD5 Proprietary Relay Protocol Avoiding VoIP Blockings 1 3 Hardware Features ARM processor DSP for voice signal processing Two 10 100MB Ethernet ports IEEE 802 3 standard with status LEDs Quadband GSM module 850M 900M 1800M and 1900M External Antenna Internal Antenna option for selected models 1 4 Software Features LINUX OS Built in Web Server for dev...

Page 6: ...in case the unit is to be transported in the future Please inspect the shipping container and the contents for any damages If visible damages are present please contact your vendor Keep the shipping materials for the carrier inspection The package should contain the items listed in the table below Item Appearance Description 1 GoIP 1 Channel GoIP 4 4 Channel GoIP 8 8 Channel GoIP 16 16 Channel GoI...

Page 7: ...is working normally or not 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 ...

Page 8: ...ach GSM module individually via its built in 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 car...

Page 9: ...N 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 inside ...

Page 10: ...the GSM channels available 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 The GoIP will then return back the LAN port IP address Please refer to 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...

Page 11: ...ule can access the designated SIM card successfully N means unable to access the designated SIM card Please check if the SIM card is inserted properly or the SIM card is damaged If Remote SIM function is used please check the SIM Bank and or SIM Server configuration The problem could also be caused by bad network condition or improper network configuration 4 GSM GSM registration status Y means Reg...

Page 12: ...increment in rssi values means 2 dBm increment 99 means that the signal level is unknown or undetected 9 Carrier This shows the name of the current GSM carrier 10 Cell ID This shows the Base Transceiver station BTS ID 11 Idle This shows the time elapsed since the last call 12 Remain This shows the time remaining if the Total Talk Time Limit m is set Once the Remain time reaches zero the correspond...

Page 13: ...r as a one to one relationship For outgoing calls from VoIP to GSM the GSM channel selection is based on the Routing Prefix VoIP a Line This is used as a reference in VoIP line configuration b Mode This shows the current VoIP Registration mode S means Single Server Mode L means Config By Line mode Gx means Config by Group mode where x is the group reference number T means Trunk Gateway mode c Logi...

Page 14: ...ters to the GSM network successfully 4 RSSI Received Signal Strength Indicator Please see the description in Section 3 1 1 5 GPRS Login Y means access to a GPRS network This status is obtained from the command AT CREG 6 GPRS Attach Y means GPRS Attach is successful and is ready for PDP This status is obtained from the command AT CGATT 7 Carrier This shows the name of the current GSM carrier 8 GSM ...

Page 15: ...the current call forward settings of the SIM card assigned to the corresponding channel There are 3 possible status 1 ON This means that the corresponding Call Forward mode is enabled and this setting is sent to the GSM network when a new GSM registration takes place 2 OFF This means that the corresponding Call Forward mode is disabled and this setting is sent to the GSM network when a new GSM reg...

Page 16: ...ration on the left hand column to display the Configuration page and the following submenu 1 Preference 2 Network 3 Basic VoIP 4 Advance VoIP 5 Media 6 Call Out 7 Call Out Auth 8 Call In 9 Call In Auth 10 SIM 11 SIM Forward 12 IMEI 13 SMS 14 GSM Carrier 15 GSM Base Station ...

Page 17: ...ge This sets the webpage and voice prompts language Currently only English and Simplified Chinese Mandarin for voice prompt are supported English 2 Time Zone This specifies the offset of the local time zone with respect to GMT The syntax should be GMT x where x is the offset 3 Time Server This specifies IP address or the domain name of a network time server for computer clock synchronization The d...

Page 18: ...Server administrator for the communication port This specifies the name to be appeared in the Remote Control Server It is used as a reference for the device This specifies the login password to the Remote Control Server This is not the password to login to the built in webpage Please ask your Remote Server Administrator if it is not available 1920 6 Network Tones Network tones are the tones associ...

Page 19: ... Service Center Fill in the SMPP ID Password and port number for SMPP data communications 13 DTMF Tone Min Gap 200 400 This parameter specifies the maximum dropout time for a DTMF tone When making a call from SIP to GSM or from GSM to SIP by using the second dial method the device needs to detect the dialing digits from the DTMF tones received via the voice data stream Depending on the network con...

Page 20: ... the device in order get network connection It 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 DH...

Page 21: ...t you send us your application requirements in full details when seeking for technical support in configuring the device In order to simplify SIP configuration SIP settings are categorized as Basic VoIP Advanced VoIP and Media In general Basic VoIP defines how the GoIP handle SIP calls and four SIP modes are supported It is important to understand the differences between each mode in order to sele...

Page 22: ... Routing Prefixes are set An idle channel is selected to dial out the call If no idle channel is available the call is rejected by sending back a SIP 503 message It is important to note that the Routing Prefix P must be removed via the dial plan before the number is dialed out The dial plan syntax to remove the Routing Prefix is P P Please refer to section 3 3 6 for more information on the Call Ou...

Page 23: ...de is only applicable for multi line models Each line associated with a corresponding GSM channel registers to a SIP server separately and operates as an independent phone line A Routing Prefix for each channel must be assigned in order to enable the channel to allow making outgoing calls This allows to the SIP server to assign which channel to dial out the call for termination If a channel does n...

Page 24: ...oup for 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 t...

Page 25: ...twork segment it is recommended that both SIP server and the device are using public IPs for reliable operation Installing either one or both SIP server and the device behind NAT may or may not work properly depending on the SIP Server and the routers on each side as well In this case SIP server must support NAT Some routers may map an internal port to a different external port number Otherwise Vo...

Page 26: ...r Period must be set to zero The parameters available 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...

Page 27: ... parameter specifies the response to the INVITE message The three possible responses are described in details below 1 SIP 200 OK This response inform the SIP server that the call is answered and the call duration timers starts immediately If billing applies the call is charged immediately even before the call is answered 2 SIP 180 then 183 The device first sends back a SIP 180 Ringing to the calli...

Page 28: ...reate 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 SIP registration to the built in SIP server Disabled 5 NAT Keep Alive When enabled NAT Keep Alive sends a NULL packet to the router regularly in order to keep the network port...

Page 29: ...ly so that they are compatible with the SIP Server and your requirements 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...

Page 30: ...GoIP User Manual http www dbltek com 29 3 3 5 ...

Page 31: ...eters Media Description Default Value 1 RTP Port Range This specifies the range of RTP port to be used for audio stream 16384 32768 2 Packet length ms This specifies the length in time of each packet However the packet length is codec dependent as well The minimum packet length of a codec supersedes the valued specified here The table below summarizes the possible packet length for the codec suppo...

Page 32: ...ode This specifies the maximum jitter delay for Adaptive Jitter Buffer mode Fixed 60 4 Media QoS Similar to Signaling QoS this parameter enables the QoS property for audio packets Both IP ToS Type of Service and DiffServe Differentiated Service method are supported None 5 Media Encryption Similar to Signaling Encryption item 6 in this table this parameter enables the encryption for audio packets E...

Page 33: ...or website http www bandcalc com Place a tick mark in the check box enable the corresponding codec The codecs are listed in a descending order of priority for codec selection This means that the top one in the table will have the highest priority to be selected when establishing a call To change the priority select the desired codec and the click on UP or DOWN button on the left Note The effective...

Page 34: ...led out The Auto Config Other Line button is provide to facilitate the programming of each channel After the parameters for CH1 are set clicking this button automatically duplicates the same settings to all other channels Please note that how a VoIP call is routed to a GSM channel depends on the Configuration mode selected Please refer to the Basic VoIP section for more information The parameters ...

Page 35: ...ank 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 Config by Line mode and Config by Group mode since SIP registration is required If this parameter is blank and the Callee Number does not equal to the SIP Number defined for this line GoI...

Page 36: ...er defines how incoming VoIP calls are authenticated before dialing them out via the GSM network Five options are available 1 None No authentication is required Calls are always dialed out via the GSM network 2 Password The caller is prompted for entering the password before the call is dialed out 3 Whitelist This is part of the Call Screen function Only the caller numbers listed on the Whitelist ...

Page 37: ...f one GSM channel to Host and all other channels to Client Please do not set more than one channel to Host as it may cause the Hunt Group mode not to function properly In addition the GSM Forwarding for all Client channels must be disabled 4 The Call IN Auth is used authenticate the incoming calls Calls can be accepted or rejected based on this setting and the corresponding White list or Black lis...

Page 38: ... in the Preference section is enabled the device generates a voice prompt to ask the caller to dial an extension number otherwise it generates a second dial tone If a phone number is assigned to this parameter a SIP INVITE to the Forward Number is sent to the SIP Server or the SIP trunk address This number must be a number that can be recognized and accepted by the VoIP network registered This mea...

Page 39: ...starting prefix of 13 and a length of 11 digits Once this condition is met the number will be dialed out immediately The maximum length for the Dial Plan definition is 140 ASCII characters There is no limit on the number of rules defined Each rule must be ended with the character The rule matching starts from the beginning and stops once a match is found 5 Hunt Group Mode Forward Mode Hunt Group o...

Page 40: ...ts the number of consecutive calls with the same Caller ID required to trigger the Incoming Call Block function Disable 3 3 10 Call IN Auth This page defines how incoming calls to each GSM channel are handled GSM incoming calls can either be answered and prompted for second dial or forwarded to a phone extension or IVR in the VoIP network connected The parameters in the Call In Page are divided in...

Page 41: ...onfigured as the server in HUNT Group mode 1 Unconditional Call Forward Incoming calls are always forwarded to an idle channel If all Client channels are in use the Host channel answers an incoming call 2 Call Forward Busy The Host channel only forward in incoming call when the line is busy 3 3 11 SIM This section contains a set of parameters are related to the SIM Card of the channel selected The...

Page 42: ... allowed as per the period defined When the accumulative talk time since the period starts reaches the Talk Time Limit defined the SIM card is disabled for outgoing calls until the period expires If there is an overlap between two periods defined the first period takes priority The next period starts from the end of the first period If there is a gap between two periods the talk time for this gap ...

Page 43: ... the caller or not Enabling this parameter hides My GSM number from the called party This specified if the phone number of the caller is shown at the called party or not 3 3 12 SIM Forward This section allows the user to define the GSM Call Forward settings for each channel There are four Call Forward conditions 1 Always Forward all incoming calls unconditionally 2 Busy Forward all incoming calls ...

Page 44: ...e forwarded calls under this condition Not Set 4 Call Forward Unreachable Forward Num Forward calls when the GSM channel cannot register to the carrier This specifies the phone number to receive forwarded calls under this condition Not Set 3 3 13 IMEI This section allows IMEI modifications for all Channel in a single page At the bottom of the IMEI list user can also enable an option to change the ...

Page 45: ...different modes of operations in order to meet the different requirements from various SIP servers Please note that the SIP server registered must be configured for this operation 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 o...

Page 46: ...ch the port value set in the SMS server This specifies the login ID for the channel selected SMS Client ID and password must first be created in the SMS Server This specifies the login password for the SMS Client ID 2 Validity Period This specifies the message expiration time in the Message Center When this period expires the undelivered message is discarded The Validity Period VP is an integer fr...

Page 47: ...vider is not selected based on the base station signal strength The GoIP may then register to a GSM service provider that charges for expensive roaming fees In order to avoid this the Fixed mode should be selected in order to lock the channel to a preferred service provider When the Fixed mode is first selected you must press Save Changes to save the setting and then refresh the browser after a fe...

Page 48: ...ere are three ways to define how to switch to a different base station If it is disabled the base station selection cycles through the neighbor list If the Whitelist authentication mode is selected then the channel selection cycles through the base stations defined in the Whitelist If the Blacklist authentication mode is selected then the channels listed in the Blacklist will not be selected 4 Whi...

Page 49: ...Tools on the left hand menu to access the submenu as shown below Please note the available options under the Tools menu 3 4 1Online Upgrade Click Online Upgrade to upgrade the device firmware The current version is displayed as well as the last upgrade time ...

Page 50: ...ted If this happens pleases contact technical support for assistance c Please reboot the device if an upgrade attempt fails before performing another upgrade 3 4 2Change Password Click Change Password to change the password with respect to the login level There are three login levels 1 Administrative Level Login ID is admin and the default password is admin 2 User Level Login ID is user and the de...

Page 51: ...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 later the service provider sends back a USSD message response as shown below Click Back to return to the Send USSD command page For certain service requests user response...

Page 52: ...In Box Click SMS In Box to view the SMS messages received as shown below Select the desired line to view the latest 5 messages received for the corresponding GSM channel 3 4 6GSM Channel Control This feature is implemented for two functions 1 Removing the power to a GSM channel before removing or inserting a SIM card This is the recommended procedure in order to prevent damages to the SIM card 2 D...

Page 53: ...on If a password is required when restoring a saved configuration enter a password before the backup To restore a saved configuration choose the configuration file in the Restore Configuration section and then click Restore Enter the password if required 3 4 8Reset Click Reset to reset the device configuration back to the factory default Click OK in the pop up window shown below to confirm this ac...

Page 54: ...GoIP User Manual http www dbltek com 53 3 4 9 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 ...

Page 55: ...w summarizes 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 passwo...

Page 56: ...parties via its own phone network The device may or may not take part in the actual call 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 1...

Page 57: ...E Contact sip 20001 192 168 2 237 5060 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 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 w...

Page 58: ...o lines in the SIP MESSAGE command Example SMS SIP Recipient 3999 SIP Proxy 192 168 2 1 GSM SMS Sender 861361234567 GSM SMS Content 075583185700 SIP MESSAGE Sent to SIP Server MESSAGE sip 3999 192 168 2 1 SIP 2 0 Via SIP 2 0 UDP 192 168 2 162 5060 branch z9hG4bK1967685528 From sip 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 20...

Page 59: ... sent out to the phone number 1368266800 via the GSM channel associated with 2001 SIP SMS Sender 3999 SIP SMS Recipient 2001 SIP SMS Content 13682626800 Hello world SIP MESSAGE Sent from the SIP Server MESSAGE sip 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 19...

Page 60: ...epeated 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 durati...

Page 61: ... 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 ...

Page 62: ... in a SIP 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...

Page 63: ...oIP are 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 dete...

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