3.3.2 Gateway Configuration
System Parameters
General
- Saving call data (CDR) – select the call types on which records are to be saved into the CDR file.
- Gateway ID – identifies 2N® StarGate / BlueTower numerically in the CDR in case multiple devices generate the CDR in the network.
- General selection of SIM – select the cards for the SIM card holder to be used by the gateway. This parameter has no influence on the gateways connected to the 2N® SIM Star system.
- According to GSM groups – select the SIM card according to the settings of the Outgoing GSM group to which the wireless module is assigned.
- SIM card in slot # – hardware selection of the SIM card holder position. This setting overrides the Outgoing GSM group setting!
- Number for remote control (PRI 1, PRI 2) – enter the remote control number for data calls via the B-channel of the ISDN PRI 1 or PRI 2 interfaces.
Automatic Logout of GSM Modules
This function is used for automatic logout of wireless modules from the wireless network in a selected time interval. If the logged-in modules are occupied by a call, the logged-out modules log in automatically. If the traffic drops, the modules log out again from the network at random intervals.
- Enable automatic logout – enable/disable the feature.
- Logout hour – set the time interval for the wireless modules to start logging out randomly from the wireless network.
- Login hour – set the time interval for the wireless modules to start logging in back to the wireless network.
- Min. logged-in modules – set the minimum count of wireless modules that remain logged-in.
Summer / Winter Time
- Automatically switch to summer/winter time – enable an automatic change of the gateway system time at the time of transition to/from the daylight saving time.
- Date of switch to winter time [dd.mm] – transition day and month.
- Date of switch to summer time [dd.mm] – transition day and month.
System Restart
- Enable system restart – enable/disable the feature.
- Time of system restart [hh:mm] – set the time for system restart.
Note
- If there are active calls, restart will be executed 10 seconds after the end of the last call.
Others
- PIN – PIN code for the SIM cards that request the PIN code. Caution
Tip
- A SIM card with a PIN code other than that set in the GSM gateway configuration will be blocked with the „pin-err‟ cause. To unblock the so-rejected SIM card, enter the correct PIN on your mobile phone!
- End of dialling (empty=off) – a selectable DTMF code for DTMF dialling end in the event of a DISA incoming call. The default value is „#‟.
List of Emergency Numbers
The window displays a list of emergency numbers, which are normally routed to the PRI interface. If the PRI line is disconnected, the emergency numbers are dialled automatically via any GSM/UMTS module according to the following rules:
- Search of a logged-in GSM/UMTS module (regardless of free minutes);
- Search of a blocked or network searching GSM/UMTS module.
The table includes an exact format of the number to be called (112,911, etc.). The ‘x’ placeholder stands for any digit in the number to be called. The ‘_’ placeholder means the rest of the number. For example:
Format | Allowed numbers |
---|---|
123 | 123 only |
14x0 | 1400, 1410, 1420, ...1490 |
999_ | All numbers starting with 999 |
LED Indication
- GSM signal mode – set the GSM module signal LED indication.
- None
- Module1 only
- Module2 only
- All modules
VoIP Parameters
VoIP Functions
- Day of deleting statistics on VoIP (every month) – set the day for automatic deletion of call statistics via the VoIP interface. None = statistics will not be deleted automatically.
SIP Settings
- Use CLIP from INVITE field – define that CLIP from the ‘Contact’ or ‘From’ field shall be used for call routing to GSM/UMTS.
- Send 180 ringing instead of 183 session progress.
- Send 200 OK instead of 180/183.
- Send 200 OK and BYE when rejected from GSM.
- Send 200 OK on REGISTER request – virtual registration of device in 2N® StarGate / BlueTower (for registration requiring equipment).
- Replace CLIP from GSM with Caller ID.
- Deny DTMF according to RFC2833.
- Use SIP over TCP for registration.
- Use SIP over TCP for calls.
SIP Registration
- Registration expires [s] – set the expiration time for the 2N® StarGate / BlueTower registration data with SIP proxy.
- Reattempt registration [s] – set the time interval after which the request shall be resent.
- Registration domain (realm).
- Caller ID.
- Username – registration data with SIP proxy.
- Password – registration data with SIP proxy.
Voice Parameters
- First RTP port (even: 1024 – 65524) – set the number of the first RTP port. The RTP port number must be even according to the recommendation.
- Last RTP port (even: first RTP+10 – 65534) – set the number of the last RTP port. The RTP port number must be even according to the recommendation. The recommended minimum RTP port range is 10.
Codec Settings
Additional settings for the voice codec used. For the default settings see the table below.
Codec | Number of blocks | VAD (Voice Activity Detection) |
---|---|---|
G.711 | 4 | Deactivated |
G.729 | 2 | Deactivated |
G.723 (6.3 kbps) | 2 | Deactivated |
Caution
- Using low rate codecs may deteriorate voice quality!
- You have to have a wide-band TCP/IP connection for using a high rate codec!
- You can use an Internet VoIP bandwidth calculator to calculate the actually required bandwidth (e.g. www.bandcalc.com).
Codec Priority
Set the types of the speech codecs to be preferred.
- Priority 1
- Priority 2
- Priority 3
- Priority 4
IP Addresses
- VoIP board – required IP address of the VoIP card of the connected gateway.
- SIP proxy (IP / GSM) – the SIP Proxy IP address from which 2N® StarGate / BlueTower awaits the GSM outgoing call requests.
Tip
- If you keep the default values (0.0.0.0), 2N® StarGate / BlueTower will receive requests from any IP address.
- SIP proxy (GSM / IP) – the SIP Proxy IP address to which 2N® StarGate / BlueTower turns in the case of GSM incoming calls.
- SIP registrar – SIP registration server IP address.
Tip
- You can use the domain name Registration domain (realm) for the SIP proxy (IP / GSM), SIP proxy (GSM / IP) and SIP registrar IP addresses on condition that you complete the domain name Registration domain (realm) and set the DNS server address properly in the Web configuration / Ethernet configuration section. The SIP proxy and SIP registrar IP addresses must be set to the default value (0.0.0.0).
- NAT firewall – NAT firewall IP address.
- STUN server – STUN server IP address (Simple Traversal of UDP through NATs (Network Address Translation)) for obtaining the public IP address with which 2N® StarGate / BlueTower operates in the Internet. You are recommended to complete this field if 2N® StarGate / BlueTower is installed in a private network separated from the Internet via NAT or firewall. The pre-set port for sending requests to STUN is 3478.
- Next STUN request (60–6553, 0=off) [s] – update of information on the 2N® StarGate / BlueTower public IP address. Use this parameter to configure the frequency of queries routed to the STUN server.
Note
- In case the GSM gateway is installed behind the NAT, make proper routing settings in the NAT router for the relevant ports (SIP, RTP, STUN). Integrated firewalls can also affect VoIP calls!
Tip
- Should you have call troubles (such as unilateral audibility, connection errors), make sure that all the active elements on the VoIP call route have been set properly. For easy troubleshooting, try the point-to-point connection with the software IP phone (SJ phone, e.g.) in your PC and, at the same time, apply network analyser tracing (WireShark – www.wireshark.org).
- Refer to the 3.3.7 Others subsection for easy tracing by 2N® StarGate / BlueTower.
Tones Generated to VoIP
- Ring tone to VoIP – generate a ringing tone of your own, or transmit a real ringing tone from the GSM/UMTS networks.
ISDN Parameters – General
Use this window to set the PRI ISDN port parameters.
PRI Mode Selection
Here define the PRI 1 (2) ISDN port type.
- Synchronisation – define that the gateway will send the synchronisation clock (MASTER), or receive the synchronisation clock from the connected line (SLAVE).
PRI1 / PRI2
- Channel number select – define the way of occupation of B-channels on a selected ISDN PRI interface. With the UPWARDS selection, the gateway will occupy the B-channels upwardly (from the selected B-channel number to the B-channel 32). With the DOWNWARD selection, the gateway will occupy the B-channels in the descending order (from the selected B-channel number to B-channel 1).
- Preferred channel – select a B-channel.
Note
- The PRI 1 port is always set as internal – the LCR function is activated. The PRI 2 port is always set as external – all calls from PRI 2 are rerouted to PRI 1. The PRI 2 port has a mode opposite to that of the PRI 1 port.
- In the case of ISDN port mode change, change the selection of the ISDN PRI card jumpers, see the PRI Board subsection.
PRI1 and PRI2
- TEI address – Terminal Equipment Identification number for the ISDN PRI line connected. The TEI must be identical for both devices on one ISDN PRI line. The default value is 0.
- CRC – enable/disable receiving CRC on Layer 1 on the ISDN PRI interfaces (in both modes the gateway sends the CRC).
Progress Indicator Value
Define which progress element value will be sent in the selected ISDN messages. The correct setting of these parameters depends on the current settings of the opposite device (PBX / Router). This parameter is primarily designed for opening the B-channel before call connection – transferring call state tones (e.g. ringing tone) from GSM / UMTS networks. The following table contains common values of progress elements.
Value | Meaning | Event |
---|---|---|
0 | No progress element sending | B-channel closed |
1 | Call is not end-to-end ISDN or may be in-band information | B-channel opened |
2 | Destination call address is non-ISDN | B-channel opened |
3 | Origination call address is non-ISDN | B-channel opened |
4 | Call has returned to the ISDN | B-channel closed |
8 | In-band treatment has been applied | B-channel opened |
10 | Delay in response at the call interface | B-channel closed |
Caution
- Wrong settings of these parameters may cause malfunction of the ISDN PRI line (call connection)!
PRI Functions
- Day of deleting statistics on PRI (every month) – set this value to enable deletion of statistic data on x-th day of a month.
- Digits count in SETUP (en-block) – set the number of digits to be sent in the SETUP message as ENBLOCK. The others digits will be sent in the OVERLAP dial mode. Example of use:
Called party number: 6012345678, digits count in SETUP = 7
Outgoing signalling messages: SETUP (contains 6012345)
INFO (6)
INFO (7)
INFO (8)
- Receive dial number from Subaddress – with this parameter activated, the gateway uses the digits in the Subaddress field instead of the Called party number for outgoing dialling.
- Hungarian dial tone after 2 digits – a specific feature designed for the Hungarian national dialling plan. The gateway activates a specific dialtone for second dialling after receiving two digits.
- Don't send ConnectAck on TE – when the ISDN PRI is in the TE mode, the gateway will not send the CONNECT_ACK message.
- Use CLIR if requested from ISDN (SETUP) – enable automatic CLIR resending to GSM/UMTS if required so by the ISDN.
Tone Signalling for Calls from ISDN
- Dial tone to PRI1 with empty SETUP – set the dial tone type to be generated by the PRI gateway.
- Ring tone – set the ring tone type to be generated by the PRI gateway.
- Generate busy tone to PRI1 – set the busy tone type to be generated by the PRI gateway into the PRI 1 interface.
- Generate busy tone to PRI2 – set the busy tone type to be generated by the PRI gateway into the PRI 2 interface.
- Ring tone from PRI1 to PRI2 – set the ring tone type to be generated by the PRI1 interface into the PRI 2 interface.
Numbering Plan Settings
- CDN, CGN – use these parameters to set the Numbering plan for the called (CDN) and calling (CGN) numbers.
Binary value | Decimal value | Description |
---|---|---|
0000 | 0 | Unknown numbering plan |
0001 | 1 | ISDN/Telephony numbering plan |
0011 | 3 | Sata numbering plan |
0100 | 4 | Telex numbering plan |
1000 | 8 | National standard numbering plan |
1001 | 9 | Private numbering plan |
1111 | 15 | Reserved for Extension |
ISDN Parameters – Monitoring
Caution
- This service is subject to licence! Refer to Management/Licence key for the current GSM gateway licence state details.
PRI1 Alerts
Send SMS at state changes – enable/disable alert sending upon PRI1 state change.
PRI2 Alerts
- Send SMS at state changes – enable/disable alert sending upon PRI2 state change.
PRI Common Settings
- Timeout for ISDN line deactivation detection [s] – an SMS alert on PRI1/PRI2 deactivation is sent after this timeout.
- Timeout for ISDN line activation detection [s] – an SMS alert on PRI1/PRI2 activation is sent after this timeout.
- Numbers where SMS will be sent to – list of numbers to which the SMS alert shall be sent.
- Text of SMS – SMS alert text with the following parameters:
- %P(x|y), where 'x' represents any text concerning PRI1 and 'y' represents any text concerning PRI2.
- %A(x|y), where 'x' represents any text for interface deactivation and 'y' represents any text for interface activation.
Alive SMS Interval Settings
- Send Alive SMS – enable/disable sending of ALIVE SMS.
- Time [hh:mm] – Alive SMS sending time.
- Days interval – Alive SMS sending interval.
- Numbers where SMS will be sent to – list of numbers to which the Alive SMS shall be sent.
- Text of SMS – Alive SMS text.
GSM Basic Parameters
GSM Selection
- Assignment of GSM channel
- Cyclical
- Locked – peer to ISDN channel
- Smart – least used minutes
- Smart – most remaining minutes
- Linear – always first free module
Caution
- The Locked – peer to ISDN channel selection has a higher priority than the LCR table. All outgoing/incoming calls are locked in the GSM-ISDN channel pair. The GSM channel is selected cyclically for outgoing VoIP calls.
- The Smart – least used minutes and Smart – most remaining minutes rules are useful where different limit settings are used for SMS.
Number of Digits Dialled from ISDN
- Minimum digits – set the minimum count of digits to be dialled into the GSM network.
- Maximum digits – set the maximum count of digits to be dialled into the GSM network.
- Wait for next digit [s] – set the timeout for 2N® StarGate / BlueTower to wait for the next digit dialled from ISDN to GSM.
Calls
- Relax timeout [s] – set the timeout between the end of one and beginning of the next call via one and the same GSM module (incoming and outgoing calls are rejected during this timeout). The recommended relax value is 2 seconds – please do not change this setting unless absolutely necessary.
- Timeout for ringing to GSM [s] – set the ringing timeout for outgoing calls to GSM. If not answered or terminated within this timeout, the call will be terminated automatically by the gateway when this timeout elapses.
- Source interface for CallBack – select the CallBack source interface for CallBack routing. The LCR table must include the appropriate outgoing traffic rule: if, for example, a VoIP port is the CallBack source interface, routing from VoIP to GSM must be defined in the LCR.
- Delay for fast CallBack [s] – set the delay between the CallBack request and outgoing call if Auto end to CallBack request is active.
Holiday List
List of days on which calls will be routed like on weekends in the LCR.
DTMF Settings
Set the minimum delay between two identical DTMF characters [s/100] received.
Tone Detector Settings
The GSM gateway can automatically detect user defined tones transmitted by GSM/UMTS for setting up outgoing calls to the GSM/UMTS networks. In general, these are tones of a transferred number. For the purpose of such detection, the GSM gateway automatically terminates the call and tries to set it up via another available outgoing group (if defined in the LCR).
- Frequency 1;2;3;4 – define the frequencies for the tone to be detected.
- Sequence list – set the sequence of the tones to be detected.
Voice Message Detector Settings
- Minimum percent to match – set the match percentage range in which the voice message is detected as identical with one of the voice messages recorded in the Gateway control/Voice messages section under index 30–37. The recommended value is 70–90%.
Ericsson Module Settings
- Transmission volume [dB] – set the module volume for the outgoing direction.
- Reception volume [dB] – set the module volume for the incoming direction.
Cinterion Module Settings
Transmission volume [dB] – set the module volume for the outgoing direction.
Reception volume [dB] – set the module volume for the incoming direction.
Enable connection tone – enable the connection tone.
End call with SHUP – end calls with the SHUP command.
Enable HR codec – enable the Half Rate codec for the GSM network.
Enable AMR codec – enable the Adaptive Multi-Rate codec for the GSM network.
Echo cancelling (MC55) – enable/disable echo cancelling for Cinterion MC-55.
Wavecom Module Settings
- GSM-band selection – select the GSM frequencies for the mobile network.
- Echo cancelling – enable/disable echo cancelling for the module.
Transmission volume [dB] – set the module volume for the outgoing direction.
Reception volume [dB] – set the module volume for the incoming direction.
Enable HR codec – enable the Half Rate codec for the GSM network.
Enable AMR codec – enable the Adaptive Multi-Rate codec for the GSM network.
Sierra Wireless Module Settings
Transmission volume [dB] – set the module volume for the outgoing direction.
Reception volume [dB] – set the module volume for the incoming direction.
- Type of used networks – select the network types for the module to log in.
- Searching priority – set the network type searching priority.
- Init module after call – initialise the module settings after call end.
- Echo canceller mode – enable/disable echo cancelling for the module.
- Enable noise reduction – enable/disable noise reduction for the module.
Telit Module Settings
- Type of used networks – select the network types for the module to log in.
- Automatic band selection – enable the automatic mobile frequency band selection.
- GSM-band selection – select the GSM frequencies for the mobile network.
- UMTS-band selection – select the GUMTS frequencies for the mobile network.
Enable HR codec – enable the Half Rate codec for the GSM network.
Enable AMR codec – enable the Adaptive Multi-Rate codec for the GSM network.
- Enable FR AMR Wideband codec – enable the Full Rate Adaptive Multi-Rate codec for UMTS.
- Enable UMTS AMR Version 2 codec – enable the Full Rate Adaptive Multi-Rate version 2 codec for UMTS.
- Enable UMTS AMR Wideband codec – enable the Adaptive Multi-Rate WideBand for UMTS.
- Enable echo canceller – enable/disable echo cancelling for the module.
- Enable noise reduction – enable/disable noise reduction for the module.
Warning
- The GSM/UMTS codec and noise reduction settings of the above listed module types may affect the DTMF detection quality. Make sure that the settings comply with the mobile network used.
Audio Level
Output audio level of GSM codec [dB] – audio volume gain/loss to GSM.
Input audio level of GSM codec [dB] – audio volume gain/loss from GSM.
Tones Generated for Incoming Calls from GSM/UMTS
- Dial tone – set the dial tone for incoming calls from GSM/UMTS.
- Ring tone – set the ring tone for incoming calls from GSM/UMTS.
- Generate busy tone to GSM/UMTS – enable busy tone generation for call termination.
Caution
- If the Generate busy tone function is enabled, the duration of outgoing calls billed by the GSM/UMTS provider will be extended!
Error GSM/UMTS Causes
Set the ISDN release cause for each of the below mentioned statuses. Every call that meets any of the below mentioned requirements, will be rejected with a user defined cause (the ISDN cause number will be translated to VoIP as a SIP code as defined below).
- Lack of digits in OVERLAP mode – any call that fails to meet the minimum digits count requirement will be rejected.
- Restricted number prefix – any call whose prefix is not included in any of the prefix lists will be rejected.
- Selected module / GSM group is not ready – a call will be rejected in case there is no available GSM module in the selected (by LCR) outgoing GSM group.
- Selected module / GSM groups are not ready – a call will be rejected in case there is no available GSM module in the selected (by LCR) outgoing GSM groups.
ISDN cause value | Description | SIP code | Description |
---|---|---|---|
1 | Unallocated number | 410 | Gone |
3 | No route to destination | 404 | Not found |
6 | Channel unacceptable | 503 | Service unavailable |
16 | Normal call clearing | BYE | |
17 | User busy | 486 | Busy here |
18 | No user responding | 480 | Temporarily unavail. |
19 | No answer from user | 480 | Temporarily unavail. |
21 | Call rejected | 603 | Decline |
22 | Number changed | 410 | Gone |
27 | Destination out of order | 404 | Not found |
28 | Address incomplete | 484 | Address incomplete |
29 | Facility rejected | 501 | Not implemented |
31 | Normal, unspecified | BYE | |
34 | No circuit available | 503 | Service unavailable |
38 | Network out of order | 503 | Service unavailable |
41 | Temporary failure | 503 | Service unavailable |
42 | Switching equipment congestion | 503 | Service unavailable |
44 | Requested facility not subscribed | 503 | Service unavailable |
47 | Resource unavailable | 503 | Service unavailable |
50 | Requested facility not subscribed | 503 | Service unavailable |
55 | Incoming class barred within CVG | 603 | Decline |
57 | Bearer capability not authorised | 501 | Not implemented |
58 | Bearer cap, unavailable at present | 503 | Not implemented |
63 | Service or option unavailable | 501 | Service unavailable |
65 | Bearer cap, not implemented | 501 | Not implemented |
79 | Service or option not implemented | 501 | Not implemented |
87 | User not member of CVG | 603 | Decline |
88 | Incompatible destination | 400 | Bad request |
98 | Invalid message | 400 | Bad request |
102 | Recover on timer expiry | 408 | Request timeout |
XXX | The other received CAU from netw. | 500 | Internal server error |
Causes of Failed Calls
Define the GSM failed call causes for the 2N® SIM Star Server. If you enter no record into the Causes of failed calls table, every non-connected call will be evaluated as a failed call.
Others
- Text of SMS at no answer – edit the text of the SMS to be sent to the calling party in case of no answer (if the function is active). The %N string will insert the CLIP received from VoIP in the SMS text.
- Text of SMS for all calls – complete this parameter to make the GSM gateway send SMS to every called party regardless of whether or not the call was successfully connected. The %N string will insert the CLIP received from VoIP in the SMS text.
- Save received SMS to – select the storage for received SMS messages.
- SIM card identification – set the SIM Id (IMSI/SCID) to be entered in the CDR.
- Disable CLIP from GSM/UMTS to VoIP – enable/disable resending of the CLIP from GSM to VoIP.
- Reject call with CHLD – enable rejection of incoming calls from GSM/UMTS via AT+CHLD (subscriber busy) instead of standard ATH.
- Network registration timeout [min] – set the timeout after which the module must log in. The login process will be restarted after this timeout.
GSM Group Assignment
Here you can assign the GSM/UMTS modules to groups. You can assign incoming and outgoing calls separately using the parameters below.
GSM Outgoing Groups
- Mode of switching SIM card – define how to use the SIM cards in the selected GSM group:
- SIM # – the SIM cards in position # will only be used in the selected GSM group. In this case, the Maximum called minutes and Disconnect call parameters are disabled.
- By time – the SIM card will be switched according to the Time of use parameter and call timer limitations.
- Random – the SIM card will be switched randomly at defined time intervals and according to call timer limitations.
- By time and limit; then SIM card 1 – the SIM card will be switched according to the Time of use parameter and call timer limitations. When all SIM cards on a selected wireless engine are discharged, the system will switch the wireless engine back to the SIM card in position 1.
- Random and by limit; then SIM1 – the SIM card will be switched randomly at defined time intervals and according to call timer limitations. When all SIM cards on a selected wireless engine are discharged, system will switch the wireless engine back to the SIM card in position 1.
- Last searched SIM (8 – all SIM will be searched) – Define the number of SIM holder positions to be checked for SIM by the gateway after the GSM / UMTS board start/restart. The SIM card holders that were not tested have the (undef) status in the diagnostics window of the web interface.
- Delay for CONNECT [s] – define a delay before sending information on the connected call after receipt from GSM.
- Minimum ring duration to send SMS at no answer [s] – set the minimum ringing time for an outgoing call to GSM/UMTS before the SMS at no answer is sent.
Note
- The SMS at no answer function works properly only if the INVITE message contains the called and caller numbers.
- Delay for ALERTING [a] – define a delay before sending information on ringing start.
- off – no Alerting message will be forwarded.
- real – the Alerting message will be forwarded as soon as the gateway receives Alerting from the wireless network or ring tone detectors detect the alerting tone (if active).
- 1–20 – the Alerting message will be automatically sent after a predefined number of seconds after the call is dialled into wireless network.
- Minute parameter – select whether or not the GSM gateway should record the call length or count for outgoing call restriction.
- Day of deleting statistics in group (every month) – define a day on which statistics on disconnected calls should be deleted.
- SIM settings – select the settings to be used for the SIM cards in positions 5–8:
- SIM 5–8 is SIM 4 – all settings for the SIM cards in positions 5 – 8 will be the same as those for the SIM card in position 4.
- SIM 5–8 is SIM 1–4 – all settings for the SIM card in position 5 will be the same as those for the SIM card in position 1. The SIM card in position 6 will use settings of the SIM card in position 2, SIM 3 = SIM 7 and SIM 4 = SIM 8.
- Disable tone detection – deactivate the alerting tone detectors on the GSM / UMTS boards. Used for the GSM / UMTS networks that send call statuses through the signalling channel.
- Generate virtual ring tone – enable/disable generation of the virtual ringing tone to the VoIP interface.
- Call length counting – select whether the call should be counted in seconds or minutes.
- After call relax delay – set the interval between the current call termination and next call setup via the same GSM/UMTS module. 2 seconds is recommended for high-traffic installations.
Network Settings
- BTS lock – identify the BTS to which the GSM modules shall log in. Restart the selected GSM modules to execute the changes.
Caution
- The BTS lock service work with specific GSM modules only (Q55, Q24, GE910, HE910)!
- If you set a wrong BTS lock, the selected GSM module(s) will not log in to GSM.
- Network operator code (MCC+MNC) – set the local mobile network provider code manually. If you do not enter a value, the provider will be selected automatically.
- Number of registration attempts – set the count of network registration attempts if the network rejects to register the +CREG:3 SIM responses.
- Delay after registration denied (1-600) [s] – set the registration delay, i.e. the timeout within which the network may send another response after rejecting +CREG:3.
- Timeout for registration (10-600) [s] – set the maximum timeout for the module to wait for network login in the NWAIT mode.
- Next try for registration (0=off, 1-720) [m] – set the interval between relogging attempts.
- Enable USIM and SIM Application Toolkit – enable/disable the USIM and SIM Application Toolkit services for the Telit HE/GE 910 modules.
Disconnect Call
Set the rules for automatic disconnection of outgoing calls to a wireless network.
- SIM limit exceeded – automatic call disconnection when the active SIM card call limit is exceeded.
- Time limit exceeded – automatic call disconnection when the active SIM card time of use is exhausted.
- No ALERTING before CONNECT – automatic call disconnection when the gateway receives the call connect message without alerting.
Block SIM
Set the rules for SIM switching in the event of SIM limit exceeding.
- Tariff 0(...4) minutes limit exceeded – enable/disable SIM switching when the tariff 0(...4) minutes limit has been exhausted.
- SMS limit exceeded – enable/disable SIM switching when the SMS limit has been exhausted.
- Minutes limit exceeded – enable/disable SIM switching when the maximum called minutes limit has been exhausted.
- Daily minutes limit exceeded – enable/disable SIM switching when the maximum daily minutes limit has been exhausted.
Send CLIP from VoIP/ISDN to GSM/UMTS
- Transfer CLIP to GSM/UMTS – enable/disable the function.
- Separating char – define the CDN/CLIP separating character.
- Modify (‘–’ remove one digit) – change the CLIP. The ‘–’ character is used for deleting one character from the left.
Caution
- The Send CLIP from VoIP to GSM service must be supported by the GSM/UMTS provider’s network. Otherwise, the call may be rejected by GSM/UMTS!
GPRS Activation
- APN string – define the Access Point Name (APN) for GPRS connectivity.
Basic Settings
- Roaming enabled for network code – set the international network identifier for roaming purposes consisting of the following two codes:
- MCC – Mobile Country Code – country code (Czech Republic – 230);
- MNC – Mobile Network Code – GSM code (T-Mobile – 01, 0202, Vodafone – 03).
Hence, the international identification code for T-Mobile CZ is 23001. Leave this field empty to disable roaming.
Input | Note |
---|---|
<empty> | Roaming is restricted |
2300 | Roaming is restricted (minimum is five digits) |
23002 | Roaming is permitted for network with MCC+MNC code 23002 |
230XX | Roaming is permitted for network with MCC+MNC codes 23000 – 23099 |
XX001 | Roaming is permitted for network with MCC+MNC codes 00001 – 99001 |
XXXXX | Roaming is permitted for any network |
Note
- Before activating the roaming support, please check the SIM card for correct setting of the GSM / UMTS priority network list.
Caution
- Calls via roaming networks may cause additional call costs!
- CLIR – enable/disable presentation of the SIM CLIP on the called party’s telephone. CLIR is recommended for the SIM card inserted in the GSM module to avoid CallBack problems.
Caution
- The Temporary CLIP enable and Temporary CLIR enable services must be supported by the GSM/UMTS provider’s network. Otherwise, the call may be rejected by GSM/UMTS!
- Maximum number of called minutes – define the maximum count of minutes to be called within a month via the selected SIM card.
- Count of SMS messages – define the maximum count of SMS messages to be sent within a month via the selected SIM card.
- Day of restoring call limit and delete statistics – select a day in a month on which the Max count of called minutes and Count of SMS messages statistics shall be deleted.
- First count – set the length of the first pulse after which the pulse counting change starts as defined in the Next count parameter.
- Next count – set the length of one pulse in seconds after the time defined in the First count parameter elapses.
Note
- Set the two parameters above (First count, Next count) properly to count free minutes on SIM cards correctly. These parameters are used for limiting outgoing calls depending on free minutes. The CDRs contain real data.
- Day limit of called minutes – set the maximum count of minutes to be called within a day via the selected SIM card.
- AOC sending interval – set the interval to send AOC messages
Time Limits
There are two SIM use time limits in a GSM group.
Call tariffs – use this function to assign up to four independent free minute counters to a group of GSM modules (SIM cards), e.g.:
- Tariff 1 = free minutes for calls to own GSM/UMTS network.
- Tariff 2 = free minutes for calls to other GSM/UMTS networks.
- Tariff 3 = free minutes for calls to fixed network.
- Tariff 4 = free minutes for calls within a closed user group (VPN).
Complete the LCR table (assign prefixes to tariffs) properly to make full use of this function. If routing to a tariff is not used, the global free minutes function will be used in the LCR table.
- Free minutes – set the free minutes for the tariff offered by the provider.
- Transferred minutes – set the maximum count of free minutes to be transferred to the next period.
- AOC sending interval – set the interval to send AOC messages.
- Day of restoring free minutes – set a day on which the free minute counters will be reset automatically. Select every 24 hours, a day in a month, or a day in a week.
- Week of restoring free minutes in month – set a week in which the free minute counters will be reset automatically. Set the week number only if the restoration takes place every other Friday in a month, e.g., or keep Every for the other cases.
Caution
The recommended free minute counter value is X-5, where X is the number of free minutes obtained from the GSM/UMTS provider. Thus, you can avoid exceeding limits.
The manufacturer is not responsible for additional call costs incurred as a result of exceeding your GSM/UMTS provider’s free minute/SMS limit.
GSM Incoming Groups
2N® StarGate / BlueTower allows you to work with four groups of incoming calls. You can set different values for each of them.
General Settings
- Mode – set how the gateway shall process incoming calls from the GSM network.
- Reject incoming calls – all incoming calls from the GSM network are rejected automatically.
- Ignore incoming calls – all incoming calls from the GSM network are ignored. The calling party hears the check ring tone.
- Accept incoming calls + voice message – incoming GSM calls are accepted by the gateway and, if programmed so, DTMF with a voice message is activated for them.
- Accept incoming calls + dialtone – incoming GSM calls are accepted by the gateway and, if programmed so, DTMF with a simulated second dialtone is activated for them.
- CallBack after ring / Reject – CallBack will be made if the CLIP is included in the CallBack table. The other incoming calls will be rejected.
- CallBack after ring / Ignore – CallBack will be made if the CLIP is included in the CallBack table. The other incoming calls will be ignored.
- Report to PC + voice message – the GSM gateway sends information on the incoming call to a PC equipped with call routing application. If programmed so, DTMF with a voice message is activated for the incoming call.
- Report to PC + dialtone – the GSM gateway sends information on the incoming call to a PC equipped with call routing application. If programmed so, DTMF with a simulated second dialtone is activated for the incoming call.
- Minimum digits in DTMF – set the minimum count of digits to be requested by the gateway for DTMF.
- Maximum digits in DTMF – set the maximum count of digits to be accepted by the gateway for DTMF.
- Timeout for entering DTMF digits [s] – set the timeout for which the GSM gateway shall wait for the first/next DTMF digit. If you select ‘0’, the incoming call will be automatically connected to the numbers included in the List of called numbers.
- Day of deleting GSM group statistics – set a day in a month on which the incoming call statistics shall be deleted.
- Prefix before DISA – set a numerical prefix to precede DTMF.
- CLIP – use this parameter to modify the incoming CLIP from GSM/UMTS. For international codes, ‘+’ will be removed automatically. Use ‘–’ to remove a digit. Examples (CLIP in GSM: +420600123456):
Parameter value | CLIP to VoIP/PRI 1 | Note |
---|---|---|
Null | 420261301500 | No change in CLIP |
+ | +420261301500 | Add + before CLIP beginning |
00 | 00420261301500 | Add 00 before CLIP beginning |
–– | 0261301500 | Remove two first digits from CLIP beginning |
–––99 | 99261301500 | Remove first three digits and add 99 before CLIP beginning |
- Looping of voice message – set the voice message playback time.
Send CLIP from GSM/UMTS to VoIP/ISDN
- Transfer CLIP from GSM/UMTS – enable/disable the function.
- Separating char – set the separator for the SIM card CLIP and ID of the extension to be called.
- Modify – modify the extension ID.
Caution
- The Send CLIP from VoIP to GSM service must be supported by the GSM/UMTS provider’s network. Otherwise, the call may be rejected by GSM/UMTS.
Others
- Time to keep CLIP in table – set the record keeping time for AutoCLIP routing.
- Add record only for unconnected call – enable storing of unconnected outgoing calls in the AutoCLIP table only.
- Delete record for connected answer – enable deletion of an AutoCLIP record in the case of successful CallBack.
- Skip DTMF for numbers not in CLIP Routing table – enable this option to set DTMF to the incoming calls only whose CLIP is included in the CLIP Routing table. Make sure that the called number table includes one record at least to make the function work.
- Skip list of called numbers after failed call to wanted number – enable this function to disable forwarding of incoming calls to the numbers included in the List of called numbers if rejected after DTMF.
- Auto end to CallBack request – enable that the incoming call whose CLIP meets the CallBack settings will be rejected. If not, the call will be ignored. The CallBack function will be retrieved after call end in both the cases.
List of Called Numbers
List of numbers to be dialled if DTMF dial-in was not made. Search the table from top to bottom. If the called user is inaccessible, use the following table record.
Prefixes
Use this window to adapt the gateway to calling to various GSM providers’ networks. Set the call routing rules based on prefixes for up to 32 groups.
Prefix List 1–32
Assign up to 32 prefix groups in the LCR table.
Basic Settings
- GSM network ID – set the prefix list user code for easier orientation in the LCR.
- Default count of digits – default length of the number to be dialled into the GSM/UMTS networks for routing via the selected prefix list. Use this parameter in case the Digits count is not included in the Accepted prefixes table.
Note
- The number to be dialled to the GSM/UMTS network must meet the Count of digits condition.
- For VoIP calls, the count of digits to be dialled must be equal to or higher than the value set in the Count of digits.
- For GSM/UMTS calls by overlap dialling via the PRI NT/TE interface, the Count of digits defines the maximum count of the digits to be dialled.
- For GSM/UMTS calls by block dialling via the PRI NT/TE interface, the Count of digits is ignored.
Table of Replaced Prefixes
Use this table to replace the prefix of the received number (‘00’ with ‘+’, e.g.). You can only add or remove the prefix. This change is made before the prefix is searched for in the prefix table. Be sure to keep the '/' record in the table for a proper function.
Note
The maximum count of records in the Table of replaced prefixes is 14 for each prefix list.
The maximum table record size is 9 characters for the prefix and 9 characters for the replaced number.
Table of Accepted Prefixes
List of prefixes of called destinations to which the selected prefix list applies.
Note
The maximum count of records in the Table of accepted prefixes is 138 for each prefix list.
The maximum table record size is 9 characters.
Prefix Change for PRI 1 to PRI 2 Calls
A table of prefixes to be replaced in the case of PRI 1 to PRI 2 outgoing calls.
LCR Table
Table of outgoing Least Cost Routing (LCR) rules. Every outgoing call from the source interface is routed to the destination interface according to this table. For a call, the gateway checks the lines and if the called number prefix matches the prefix in the selected network list and the current time value is within time limitation limits, the call will be routed via the defined GSM group(s) or PRI 1/PRI 2/VoIP interface.
- From (channels/groups) – source channels or groups via which calls are routed to 2N® StarGate.
- GSM ALL – any of the GSM incoming groups can be used for call routing.
- GSM GRP1-32 – define one GSM incoming group or a range of GSM incoming groups via which call routing will be enabled.
- PRI1 (pxx,P) – any of the PRI1 channels can be used for call routing.
- PRI1 (p1-32) – define one PRI1 channel or a range of PRI1 channels via which call routing will be enabled.
- PRI2 (exx,E) – any of the PRI2 channels can be used for call routing.
- PRI2 (e1-32) – define one PRI2 channel or a range of PRI2 channels via which call routing will be enabled.
- VoIP (vxx,V) – calls are routed via the VoIP interface.
- Prefix list – prefixes to be used for a selected LCR row. Set up to 32 prefix lists.
- Prefix list 1–32 – call routing will obey the rules set in Prefix list 1–32. The count of digits to be dialled is governed by the Prefixes setting.
- All prefixes – all the prefix lists are permitted. Prefix lists 1–32 will be searched from 1 to 32. The first match will be used routing. For incoming GSM calls, the Count of digits from the GSM incoming group parameter will be applied.
- Count of digits – all the prefixes are permitted. Routing is only limited by the count of digits to be dialled.
- Time limitation – set a time validity limitation for a selected LCR row.
- Weekend usage – enable/disable a row on weekends.
- Call duration limit – set the maximum duration (minutes) for an outgoing call to the GSM/UMTS network.
- Groups – define the outgoing GSM groups or interface via which outgoing calls will be routed from 2N® StarGate. If the defined interface is inactive or the outgoing GSM group tariff is exhausted, the next row will be applied.
- GSM groups 1–32 – outgoing GSM groups. Set the tariff to be used. Refer to the GSM outgoing groups subsection for details.
- PRI1 (P) – ISDN PRI1 interface. Refer to the ISDN parameters subsection for details.
- PRI2 (E) – ISDN PRI2 interface. Refer to the ISDN parameters subsection for details.
- VoIP (V) – VoIP interface. Refer to the VoIP parameters subsection for details.
- Tariffs – select the tariff group (free minute counter) to be used for the outgoing call. Refer to the GSM outgoing groups subsection for details.
- Ignore tone detection in last group – having detected a user defined tone (refer to GSM basic settings), the GSM gateway automatically terminates the call and seeks for another call setup way. If this parameter is activated, the GSM gateway ignores the tone detection results and sets up a call when this is the only possible call establishing way.
Note
- If you use tariff routing, set the tariffs properly in the GSM outgoing groups subsection
- The maximum count of LCR table records is 64.
- The rules are applied to calls sequentially, starting from the first rule. If all the set rules are met in a row, the call is routed according to the row.
- Call routing from PRI1/PRI1, PRI2/PRI2 and VoIP/VoIP is not supported. Such calls will be rejected by 2N® StarGate.
CLIP Routing Table + CallBack
Use the table to set a fixed CLIP assignment of incoming CLIPs from GSM to the numbers of extensions to which incoming calls are routed automatically. Also, set the CLIP list in the GSM network for which CallBack is enabled.
- GSM number (CLIP) – user Id in GSM/UMTS.
- Used service
- Autodial – enable/disable CLIP routing for the selected CLIP.
- Reject call – reject call for the selected CLIP.
- Ignore call – ignore call for the selected CLIP.
- Tone dial-in – incoming GSM calls is accepted + dialtone is activated for the selected CLIP.
- DISA message dial-in – incoming GSM calls is accepted + voice message is activated for the selected CLIP.
- Dial to VoIP/ISDN – VoIP/ISDN destination number for the CLIP routing function.
- Auto CallBack – enable/disable the CallBack function for the selected CLIP according to the Gateway configuration / GSM basic settings / Calls / Source interface for CallBack setting.
- Call duration limit – set the maximum call duration (minutes).
Tip
- CallBack detects the CLIP from right to left. Thus, configure 10 rules for all the incoming CLIPs to make CallBack work properly. Each of the rules must contain one of the CLIPs: 0,1,2,3,4,5,6,7,8,9.
Note
- Remember to activate the CallBack mode in the GSM incoming groups for a proper function.
- The maximum count of CLIP routing table records is 96.
Ethernet Configuration
This window helps you configure the Ethernet interface of your gateway.
- Use DHCP – enable/disable the DHCP client function in 2N® StarGate / BlueTower.
- IP address – fixed IP address (v4) of the 2N® StarGate / BlueTower Ethernet interface.
- Subnet mask – subnet mask for the 2N® StarGate / BlueTower Ethernet interface.
- Default gateway – IP gateway IP address (v4) in the Ethernet network.
- DNS server1 – primary DNS server.
- DNS server2 – secondary DNS server. Used as a back-up when DNS server 1 is not functional.
Caution
- Incorrect settings, e.g. enabling the DHCP client service, may result in a denial of access to the 2N® StarGate / BlueTower configuration. To avoid this, restore the GSM gateway default settings. Refer to Subs. Basic CPU Board for details and Jumper for default.
Tip
- In case the gateway is in the DHCP client mode, the current assignments obtained from the DHCP server are displayed in the IP address, Subnet mask and Default gateway items.
Login Configuration
The window helps you set the access password and name for the 2N® StarGate / BlueTower web interface. Use the same data for Telnet connection too.
Caution
- Remember to change the username and password during the first gateway configuration to avoid unauthorised access!
Note
- The username and password may have up to 30 characters in total.
Web Configuration
Set additional parameters for the gateway web interface.
- Auto logout – set the count of minutes in which the current user should be logged out automatically.
- Enable web session lock – by default, one Admin user may only be connected to the GSM gateway at a moment. If another authorised user tries to log in, the previous session will be aborted automatically. By activating this function you disable such automatic logout and block any further access attempts.
- Simple login form – change the login window graphic layout. Upon activation, the GSM gateway login page will have an anonymous appearance. This function is recommended for connecting the gateway to the Internet directly.
- Use SMS user for SMS operations on the web – enable/disable the user authorised for SMS sending/receiving only.
- SMS user name/password – the gateway enables independent connection of a user authorised for SMS receiving/sending only.
Report Configuration
This window is used for setting details on automatic tracing generated by the GSM gateway.
Time Synchronisation
Set the NTP server time synchronisation.
General
- Type - enable/disable synchronisation.
- Ntp server - NTP server address and port.
- Timezone - set the time change from UTC.
Summer/Winter Time
- Automatically switch to summer/winter time - enable/disable automatic winter/summer time transition.
- Switch to summer time - set the summer time transition date and time.
- Switch to winter time - set the winter time transition date and time.