FAQ on FXS gateways FAQ on GoIPs FAQ on SIM Banks (SMB32) FAQ on EP Phones FAQ on SMS Server
FAQ on SIM Server (V1.01.1 Build 201205) FAQ (Miscellaneous)
FAQ
FAQ on FXS gateways
Can extension lines in our FXS gateways with multiple lines call each other when it is configured as Single Server Mode?
Can our FXS gateway work with all types of PSTN phones?
Does our FSX gateway support line reverse polarity?
Is FAX supported?
Is VPN supported?
The RUN LED does not flash and the Line LED does not turn on when picking the PSTN phone connected.
No voice is heard when picking up the phone?
What do you see from the LEDs and hear from the receiver of the phone when SIP registration fails?
FAQ on GoIP gateways
What is a GoIP?
What are the advantages of GoIP over FXO gateways?
Is GoIP legal?
Is GoIP certified by a test lab for international standards?
Do I need to upgrade the GoIP firmware?
How do I insert a SIM card to GoIP?
How do I insert or remove a SIM card without powering off the GoIP?
Can I send SMS via GoIP?
Can GoIP send bulk SMS?
Can GoIP receive SMS?
How do I delete received SMS messages?
What is the maximum number of characters in a SMS message?
Is GSM 03.38 Character set supported in SMS message encoding?
Why do I get line busy signal when sending USSD or SMS via the same GSM channel?
Why is there a discrepancy in the talk time measurement by a SIP Service provider and a GSM service provider?
Why can't GoIP forward a GSM call to VoIP successfully?
Can the phone number of the caller be shown on a SIP terminal for the calls forwarded from GoIP?
Can the Caller ID be modified when making an outgoing call via a GSM channel?
What is IMEI?
Is IMEI in GoIP programmable?
How do I find out the GSM signal level?
How do I interpret the GSM signal level?
Why do I hear echoes?
Why do I get poor voice performance when all GSM channels are in use?
How do I adjust the volume level?
Does GoIP keep a call log of all incoming and outgoing calls?
How do I reset a GoIP to its factory defaults?
What is PDD and what affects it?
Why do I get low ASR with GoIP?
Why do I get low ACD with GoIP?
How do I get the balance of a SIM card installed in a GoIP?
Can I limit the talk time of a SIM card?
Can GoIP only accept certain inbound and outbound calls?
Can I reject / disable all VoIP or GSM incoming calls?
Does GoIP support backup server for non-interrupted service?
Is it possible to specify which GSM channel to use for making an outgoing call?
How does GoIP select a GSM channel when making an outgoing call?
Can I configure GoIP so that it answers my VoIP call first and then let me to dial out a number via one of its GSM channels (Second dial operationa)?
Can I configure GOIP so that I can call one of its GSM channel numbers and then let me make a VoIP call (Second dial operation)?
Why is GoIP dialing out a wrong number?
Can GoIP support hunt group operation?
Can extension lines in our FXS gateways with multiple lines call each other when it is configured as Single Server Mode? (#canextensionlines)
In Single Server mode, all extension lines shares the same SIP number. Therefore, they cannot call each other. Config. By Line or Config. By Group mode should be used so that each extension line or each group has its own SIP number. In this case, calling between extension lines or groups can then be achieved.
Can our FXS gateway work with all types of PSTN phones?(#canourfxsgateway)

Not all PSTN phones work properly with our FXS gateways with the default telephone line interface settings. You may need to modify the following parameters via its built-in webpages.

a> Telephone Line Voltage - Most PSTN phones nowadays works with telephone line voltage of 48 Volts. This parameter should be enabled as shown below. If the line voltage is not configured properly, DTMF dialing may be affected as well.
b> Ringing settings - Some PSTN phones may be designed to work with certain ringing frequency and pattern. Modified these parameters shown below accordingly. Please note that the default ringing voltage is 48Volts sine wave without DC offset. Our old FXS hardware only support 48Volts square wave without DC offset. If your PSTN phone does not work because it requires higher ringing voltage or different ringing waveform, it is not compatible with our FXS gateway. Please use a different PSTN phone.
c> CID Settings - These parameter defines how CID / CLI signals are generated. Please note that only FSK based CID signals are supported. If your PSTN phone only support DTMF CID, it will not be able to display the CID of the caller. You will not to replace it with a PSTN phone that can support FSK CID signals if this CID feature is required.
CID FSK Mode defines whether the Bellcore or the ETSI FSK frequencies to be used. In the Bellcore standard, the Mark ("1") and Space ("0") frequencies are 1200 Hz and 2200 Hz. In the ETSI standard, they are 1300 Hz and 2100 Hz.
Not all CID call setup and timing parameters are support. Our FXS port only support CID signals with the call setup using 1st Ring. The duration of the first can be specified. In addition, the delay time between the end of the first ring and the generation of the CID signal can also be defined as Delay Time Before CID. The last parameter, Delay Time After CID, defines the silent period before normal ringing starts.
Does our FSX gateway support line reverse polarity?(#doesourfsxgatewaysupport)
Line reverse polarity is implemented on our FXS port when the called party answers a call. Some systems may require this line reversal as an indication to start counting the talk time for billing.
Is FAX supported?(#isfaxsupported)
Only models HT-912T and HT-922T support T.38 and G.711 Fax mode.
Is VPN supported?(#isvpnsupported)
PPTP VPN with 40-bit encryption is supported. Please make sure your VPN server is configured for 40-bit encryption. The latest models HT-912T and HT-922T are already VPN supported. For earlier models, please upgrade firmware to version 31.5-15 via the link below. http://118.142.51.162/update/A38HS-3.15-15.pkg
The RUN LED does not flash and the Line LED does not turn on when picking the PSTN phone connected.(#therunleddoesnotflash)
The FXS gateway is damaged. Please return it for repairs.
No voice is heard when picking up the phone?(#novoiceisheard)
Please make sure that the power adapter is properly connected to the FXS gateway. If the power is connected, the RUN LED should blinks. If not, the hardware is damaged. Please return it for repairs. If the RUN LED blinks, the Line LED should light up. If not, please check to see if the phone is plugged to the correct phone jack. Please make sure that the PSTN phone is working. If the Line LED still does not light up, the hardware is likely damaged, please return it for repairs.
What do you see from the LEDs and hear from the receiver of the phone when SIP registration fails?(#whatdoyouseefromthe)

If SIP registration fails, you will see that the RUN LED blinks at a fast rate and hears an abnormal tone (not a normal dial tone) from the phone receiver.

What is a GoIP?(#whatisagoip)
GoIP is the short name for GSM Gateway of VoIP. It is intended to bridge VoIP calls to GSM and GSM calls to VoIP. It is similar to a FXO gateway except that it is using the GSM network instead of the traditional PSTN network.
What are the advantages of GoIP over FXO gateways?(#Whataretheadvantages)
In principle, GoIP can achieve all the functions that an FXO gateway can do with additional values listed below.
a) With GSM costs getting lower and lower in many countries, it make more sense to use GSM for call termination rather than traditional PSTN lines. With GSM, there are no expensive charges and long waiting period for installation/reallocation.
b) You can change the phone and service provider by just changing the SIM card.
c) Text messaging via GSM SMS is powerful for developing other applications.
d) You can build your own telephone network easily.
Is GoIP legal?(#IsGoIPlegal)
Depending on your country policy, GoIPs may be considered illegal and you could be penalized if you import or use this type of products. Please check with your local authority for further information.
Is GoIP certified by a test lab for international standards?(#IsGoIPcertifiedby)
We have tested our GoIPs in house for compliance with international standards. We can offer CE Conformity Declaration for the purpose of importation. Please contact our sales for a copy if required.
Do I need to upgrade the GoIP firmware?(#DoIneedtoupgrade)
The latest firmware includes the latest bug fixes and newly added features. You can find the latest firmware releases from the URL below.
http://www.dbltek.com/news/news_lastversion.html
Please read the upgrade note for each firmware version carefully before upgrading your hardware. Incorrect upgrade may cause the hardware not to function properly. Always technical support if you have any question.
How do I insert a SIM card to GoIP?(#HowdoIinsertaSIM)

Before inserting or removing SIM cards, please remove the power to the GoIP in order to prevent damages to the SIM cards. 

The SIM card insertion method for GoIP is different from that for GoIP-4 / GoIP-8.   The diagrams below show both insertion methods. 

For GoIP, a SIM card is inserted with its cut corner in first and its metal contacts facing down.

For GoIP-4/GoIP-8, a SIM card should be inserted with its cut corner in first and its metal contacts facing up.
                               

How do I insert or remove a SIM card without powering off the GoIP?(#HowdoIinsertorremoveaSIM)

When a GoIP with multiple channels are in operation, it is not a good idea to remove its power in order to insert or remove a SIM card since other channels are affected.  It is possible to shut down the power to each GSM channel via its built-in webpage.  This page, shown below (for GoIP-4), is located under the Tools menu and the navigation link is "Tools -> MODEL CONTROL".
Select the desired GSM channel to be shut down and then click [Save].  You are now ready to insert or remove the SIM card in the selected channel.  Once completed, you can then remove the check mark at the selected channel and then click [Save].  The power to the GSM channel is then restored and normal operation is then resumed.

Can I send SMS via GoIP?(#CanIsendSMSviaGoIP)

Yes, you can send SMS via the GoIP's built-in webpage, but it is limited to a single destination (just one GSM number per message).  Please note that the sent SMS is not saved for future review.

Can GoIP send bulk SMS?(#CanGoIPsendbulkSMS)

Sending SMS via GoIP's built-in webpage is limited to a single destination (just one GSM number).  For sending bulk SMS, you need to use our SMS Server which is a free Linux based software utility.  You can find the SMS Server Installation Manual and Installation Package in the download area of our website. 

Can GoIP receive SMS?(#CanGoIPreceiveSMS)

Yes, GoIP can receive SMS and the last five messages received are kept in the SMS Box regardless of the SMS Mode selected.

How do I delete received SMS messages?(#HowdoIdeletereceivedSMS)

You cannot delete received messages in the SMS Box.  The latest 5 SMS messages are kept in the SMS BOX in the built-in webpage.  Old messages are deleted as new messages come in.

What is the maximum number of characters in a SMS message?(#Whatisthemaximumnumber)

The data length of a SMS message is 140 bytes.  By default, GoIP is using 7-bit ASCII encoding scheme and this allows 160 characters to be packed in 140 bytes.  However, if 16-bit unicode characters are entered, the encoding scheme is changed to 16-bit unicode automatically.  The maximum number can be sent is reduced to 70 characters.  Please note that the current 7-bit encoding algorithm only supports ASCII characters (excluding control characters).

Is GSM 03.38 Character set supported in SMS message encoding?(#IsGSM03.38Character)

No, it is currently not supported.  This feature is supported in the SMS Server.

Why do I get line busy signal when sending USSD or SMS via the same GSM channel?(#WhydoIgetlinebusysignal)

This problem could be fixed by using IE to send USSD and SMS.

Why is there a discrepancy in the talk time measurement by a SIP Service provider and a GSM service provider?(#Whyisthereadiscrepancy)
The discrepancy could be accounted from two possible sources.
a) SIP 183 should be enabled to start the early media. This allows the SIP Server to measure the start time of a call more accurately. Nevigation Links:configurations ----> call settings ----> Early Media Mode (Early Media&Local Ring/ Early Media)
b) Extra time delay introduced by the GoIP in reporting the end of a call to the SIP server. This delay is reduced by upgrading its firmware to the latest version. Please visit our website www.dbltek.com for the upgrade link to the latest firmware version.
Why can't GoIP forward a GSM call to VoIP successfully?(#Whycan'tGoIPforwardaGSM)

Please first check the configuration mode.  Different SIP server may accept calls forwarded by GoIP differently.  In general, SIP server accepts all calls from a SIP Trunk without SIP registration.  In this case, GoIP should be able to forward GSM calls to the SIP server successfully.  However, if SIP registration is used or is required in SIP Trunk mode, SIP server may authenticate incoming calls based on the SIP number assigned in the SIP INVITE message.  If Caller ID Forward mode is set to "Use CID as SIP Caller ID" in the GoIP, the SIP number is then replaced by the GSM CID.  In this case, the SIP server may reject calls forwarded from the GoIP.  Please check to see if the SIP server can be configured to accept calls if the SIP number field is changed to the GSM CID.  If not, you can try to check if SIP server supports Remote Party ID.  If it does, you can then change the Caller ID Forward to "Remote Party ID".  If it does not, you will have to disable the Caller ID Forward mode.  This means that the GSM Caller ID cannot be shown at a SIP client terminal.

Can the phone number of the caller be shown on a SIP terminal for the calls forwarded from GoIP?(#Canthephonenumberofthecaller)

Please refer to the answer for question 16.

Can the Caller ID be modified when making an outgoing call via a GSM channel?(#CantheCallerIDbemodifiedwhen)

No, the Caller ID of a GSM channel can either set to the number of the SIM card inserted or to "Anonymous".  This parameter is located under the SIM Card Settings in the Call Management page.

Nevigation Links: Configurations ----> call divert ----> GSM CallerID Anonymous

What is IMEI?(#WhatisIMEI)

IMEI is short for International Mobile Equipment Identity and is a unique number given to every single mobile phone.
The number consists of four groups that looks this:
nnnnnn--nn-nnnnnn-n
The first set of numbers is the type approval code (TAC). The first two digits represent the country code. The rest make up the final assembly code. The second group of numbers identifies the manufacturer:
01 and 02 = AEG
07 and 40 = Motorola
10 and 20 = Nokia
41and 44 = Siemens
51= Sony, Siemens, Ericsson
The third set is the serial number and the last single digit is an additional number (usually 0).

Is IMEI in GoIP programmable?(#IsIMEIinGoIPprogrammable)

The default IMEI can be modified via the built-in webpage.  Please see the format of IMEI described in the question 13.  Please note that the last digit is not used and should be set to 0.  Therefore, only the first 14 digits should be modified.

Please note that the IMEI field is empty.  Please insert a SIM card and power up the GoIP.  You should then see the IMEI assigned.

How do I find out the GSM signal level??(#HowdoIfindouttheGSM)

The GSM signal level is shown in the Status page under the GSM column. 

How do I interpret the GSM signal level?(#HowdoIinterprettheGSM)

The table below shows the meaning of the signal level.  It is recommended that the signal level should be 15 or above in order to get good voice quality and reliable operation.  If the signal level is too low, it may affect the ASR (Answer Seizure Rate).  If you get a poor signal reception of a GSM channel, you should try to move it to a location with better signal reception or use an external antenna with an extended cable.  Other method to improve on the antenna gain could also be used.  If you get a signal level of 99, it could mean that there is no GSM coverage or the GSM registration fails.

Why do I hear echoes?(#WhydoIhearechoes)

If echoes occur, upgrading the firmware may solve this problem.  Please visit the following URL for the latest GoIP firmware.

http://www.dbltek.com/news/news_lastversion.html

Why do I get poor voice performance when all GSM channels are in use?(#WhydoIgetpoorvoice)

Please check if there is enough network bandwidth for both upstream and downstream data traffics.  The bandwidth requirement per voice channel depends on the codec adopted and other network parameters.  Please refer to the GoIP User Manual for more information on the bandwidth requirement.
In addition, please make sure that the voice quality issue is not introduced by the GSM network in use.

How do I adjust the volume level?(#HowdoIadjustthevolume)

The volume level of each line can be adjusted via the following URL.
http://<device address>/en_US/gain.html
The volume levels of the audio streams from VoIP to GSM and GSM to VoIP 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 line / channel as well.  As a result, DTMF tones for phone dialing may not be detected correctly.  Please change these settings with great care and make sure that DTMF detections are not affected.

Does GoIP keep a call log of all incoming and outgoing calls?(#DoesGoIPkeepacalllog)

No, GoIP does not currently support the call log in its built-in webpage.  Please use SMS Server if this feature is required. 

How do I reset a GoIP to its factory defaults?(#HowdoIresetaGoIP)
There are two ways to reset a GoIP to its factory defaults.
a) Reset configuration can be done manually. 
1. Connect the power to the GoIP 
2. Press and hold the RESET button. The RUN LED will then start flashing at a fast rate after around 10 seconds. Wait till the LED stop flashing. This means that the Reset Configuration is successful.
b) Reset configuration via webpage 
1. Access the built-in webpage of the GoIP 
2. Access the Tools menu and the select Reset Config. 
3. Confirm the rest and then reboot the GoIP
What is PDD and what affects it?(#WhatisPDDandwhat)

PDD is the short Post Dial Delay.  It is the time interval between "end of dialing" by the customer and the reception (by the same customer) of the call progress signaling generated by the exchange serving this customer.  The call progressing signals can be the dial tone, a recorded announcement, or the abandon of the call. 

For call termination with GoIP, the PDD depends on the network condition, SIP INVITE Response, and GSM network.

    1. If the network condition is poor, extra network delay is introduced in the SIP communication.  This adds to part of the PDD.  In this case, the voice quality is likely to be affected as well.  Please resolve this issue promptly.
    2. SIP INVITE Response has a directly impact on the PDD.  If SIP 180 is returned, a ringback tone will be generated locally.  The caller can then hears a ringback tone while the GoIP is try to dial out the call.  This only affects on the caller's preception, but the actual amount to dial the call is not affected.

If SIP 183 is used, the PDD is then consist of network delay for SIP protocol and the delay of the GSM network.  In some case, this could be very long and the caller may hang up the call before he gets a call progress tone back from the GSM network.  To improve this, the Ringback mode in the GoIP should be set to "Local Ringing + Early Media".  The GoIP will first respond a SIP 180 to a SIP INVITE and then send a SIP 183 once call progress tones is received from the GSM network.

Why do I get low ASR with GoIP?(#WhydoIgetlowASR)
ASR is the short for Answer Seizure Rate. It is a measurement of network quality and call success rate in telecommunications. It is the percentage of answered telephone calls with respect to the total call volume. There are a number of ways that ASR can be affected.
a) Please make sure that the total number of call sent to a GoIP does not exceed the number of channels available.
b) Please make sure that all GSM channel are registered with good signal level.
c) Please make sure that the SIM cards inserted are valid and charged.
d) Please make sure that Post Dial Delay (PDD) is reasonable. Please see question 22 in this section for more explanation on PDD.
Why do I get low ACD with GoIP?(#WhydoIgetlowACD)

ACD is the short for Average Call Duration.  It tracks the length of time a customer is on the phone.  You can investigate this issue by monitoring the voice quality of all GoIP channels.  If the voice quality is poor, please try to check the GSM signal level and try to make GSM calls with a cellphone (using the same GSM network) to compare the voice quality.  In addition, the voice quality of the VoIP calls should also be verified as well.

The stability of the call connection also affects the ACD.  Please make sure that calls are not dropped by the GSM network or by the SIP server. 

The Post Dial Delay (PDD) could also affects the ACD and ASR as well.  Please see question 21 in this section for more explanation on PDD.
How do I get the balance of a SIM card installed in a GoIP?(#HowdoIgetthebalance)
Depending on the GSM Service provider, the balance of a SIM card can be obtained via the following methods:
a) Send USSD command - Please check with your GSM service provider for the USSD command to check balance. Just enter the command as shown in the diagram below via the built-in webpage. Nevigation Links: Tools ----> send USSD
b) Send SMS - Please check with your GSM service provider for the message format for getting SIM card balance. Nevigation Links:Tools --->Send SMS
c) Dial up to the GSM Service Provider - This method is included for the sake of completeness. It is not a feasible method for customers doing call termination since it is very time consuming and takes a lot of effort in managing SIM cards.
Can I limit the talk time of a SIM card?(#CanIlimitthetalktimeof)

Yes, you can set the talk time limit via the built-in webpage.  Just follow the navigation link below to access Talk Time Limit and Billing Increment.
Nevigation Links: Configuration -> Call Management -> SIM Card Settings

The Talk Time Limit specifies the maximum talk time available for the SIM card.  Once the Talk Time Limit is reached, the corresponding GSM channel is disabled.  It must be reset via the Status Page in order to activate the GSM channel again.

Billing increment is a call duration measurement unit expressed in seconds. Depending on your service provider, some services are measured and billed in sixty second increments (one minute) or the billing increment may be in durations of six or even ten seconds.  It is important to set this parameter properly in order to calculate the time usage.  Please note that the time usage calculated here may be different from the actual time charged by your GSM service provider.  Please refer to question 9 for the possible reasons for the discrepancy.

Can GoIP only accept certain inbound and outbound calls?(#CanGoIPonlyacceptcertain)

Yes, you can enable the CALL IN Authentication for inbound calls (incoming GSM calls) and CALL OUT Authentication for outbound calls (VoIP calls) and then select the "Whitelist" authentication method.  Fill in the corresponding Whitelist with the numbers (up to 15 entries) you wish to accept.  These two parameters are list in the Call Management page.

Can I reject / disable all VoIP or GSM incoming calls?(#CanIreject/disableallVoIP)

Yes, you can. 

    a) Set the CALL OUT via GSM parameter to "Disabled" to disable all incoming VoIP calls.  A SIP 503 "Service Unavailable" is returned to the VoIP server.  This prevent incoming VoIP calls from interrupting GSM to VoIP termination.

Navigation Links: Configuration -> Call Management -> CALL OUT via GSM

    b) Set the CALL IN via GSM parameter to "Disable" to disable all incoming GSM calls to the selected channel.  This prevent incoming GSM calls from interrupting VoIP to GSM termination.

Navigation Links: Configuration -> Call Management -> CALL IN via GSM

Does GoIP support backup server for non-interrupted service?(#DoesGoIPsupportbackup)

Yes, you can enable the Backup Server if you are using Single Server Mode, Config. By Line Mode, or Config. By Group mode.  When GoIP fails to register to the primary SIP Server, it will then register to the Backup Server.  For Trunk Gateway mode, you can specifies up to three Routing Gateway addresses.  Both mechanisms help to insure non-interrupted service.

Is it possible to specify which GSM channel to use for making an outgoing call?(#Isitpossibletospecifywhich)

Yes, it is possible; however, the GoIP must use Config. By Line or Config. By Group mode.  Please see the GoIP User Manual for more information.

How does GoIP select a GSM channel when making an outgoing call?(#HowdoesGoIPselectaGSM)

GoIP only requires to select GSM channels for outgoing calls when it is configured as Single Server mode, Config. By Group mode, or Trunk Gateway mode.  In order to spread out the channel usage evenly, the automatic channel selection algorithm chooses the idle channel that is used the least (in terms of the number of calls made).

Can I configure GoIP so that it answers my VoIP call first and then let me to dial out a number via one of its GSM channels (Second dial operationa)?(#CanIconfigureGoIP)

Yes, you can.  You need to the GoIP to use Single Server mode, Config. By Line mode, or Config. By Group mode and enable the CALL OUT via GSM option in the Call Management page.  Please leave the Forward Number in this section blank.  The operation described here is referred as Second Dial.

Can I configure GOIP so that I can call one of its GSM channel numbers and then let me make a VoIP call (Second dial operation)?(#CanIconfigureGOIPsothat)

Yes, you can.  You need to the GoIP to use Single Server mode, Config. By Line mode, or Config. By Group mode and enable the CALL IN via GSM option in the Call Management page.  Please leave the Forward Number in this section blank.  The operation described here is referred as Second Dial.

Why is GoIP dialing out a wrong number?(#WhyisGoIPdialing)

There are two reasons that GoIP dials out a wrong number.

    a) For call termination from VoIP to GSM, the number received is not a valid E.164 number (PSTN and GSM numbers).   Please use the Dial Plan parameter to modify the number received, so that it is a valid number.  For international calls, please make sure that the country code and/or area code are added.  In addition, you need to make sure that long distance call service is available for the SIM card inserted.
    b) For second dial operation, GoIP needs to detect the number (DTMF digits) to be dialed from the incoming voice stream.  It is possible that the number is detected incorrectly and consequently, the number dialed out is wrong.

For VoIP calls termination to GSM, the DTMF dialing method of the calling device should be set to DTMF outband signaling (RFC2833 / SIP INFO).  If RFC2833 method is used, the payload type should be matched with the server.

For DTMF detection (inband DTMF dialing) method, you can try to tune the parameter DTMF Tone Dropout.  This helps to avoid the same digit detected twice if dropouts occurs during a single DTMF tone duration.   Please refer to the GoIP User Manual for more information.

Can GoIP support hunt group operation?(#CanGoIPsupporthunt)

Yes, GoIP supports hunt group operation when it is configured to GSM Group mode.  A hunt group (or hunting) is a telephone concept that refers to the method of distributing phone calls from a single telephone number to a group of serveral phone lines.  Applications like Call Centre and Call Back Service require this type of operation so that customers only need to call a single number in order to access the service.  Please refer to the GoIP User Manual for more information.