3.4.2 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^® VoiceBlue Next numerically in the CDR in case multiple devices generate the CDR in the network.

System Restart

• Enable system restart – enable/disable the feature.
• Time of system restart [hh:mm] – set the time for system restart.

Mobility Extension (DTMF settings)

• Start dialling (quick call forwarding) – DTMF code for quick forwarding start.
• End dialling (quick call forwarding) – DTMF code for quick forwarding end.
• Hold call – DTMF code for active call holding.
• Hang up call – DTMF code for call termination.
• Follow me activation – activation of the Follow me function. The GSM gateway starts routing call to the defined GSM/UMTS subscriber number. The default value is *55.
• Follow me deactivation – deactivation of the Follow me function. The default value is #55.
• SMS at no answer activation – activation of the SMS at no answer function for a registered user. The default value is *33.
• SMS at no answer deactivation – deactivation of the SMS at no answer function for a registered user. The default value is #33. 

Others

• PIN – PIN code for the SIM cards that request the PIN code.

• 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 '#'.

VoIP parameters

VoIP functions

• Day of deleting statistics on VoIP (every month) – the day of automatic deletion of call statistics on the VoIP interface. None = no deletion.

SIP protocol settings

• Use CLIP from INVITE field – the CLIP as set in the Contact or From field will be used for routing calls into GSM/UMTS networks.
• 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 the device in 2N^® VoiceBlue Next (necessary for registration-requiring equipment).
• Replace CLIP from GSM with Caller ID.
• Deny DTMF according to RFC2833.
• Forward DTMF for ME (Mobility Extension).
  

SIP registration

• Registration expires [s] – the timeout after which the 2N^® VoiceBlue Next registration data expire at the SIP Proxy.
• Reattempt registration [s] – time interval for re-sending the request.
• Registration domain (realm).
• Caller ID.
• Username – registration data for the SIP Proxy.
• Password – registration data for the SIP Proxy.

Voice parameters

• First RTP port (even: 1024 – 65524) – number of the first RTP port. The number must be even as recommended.
• Last RTP port (even: first RTP+10 – 65534) – number of the last RTP port. The number must be even as recommended. The recommended minimum range of RTP ports should be 10.

Codec settings

• detailed codec settings (G.711a/u, or G.729).

Codec priority

•  preferential speech codec setting:
  • Priority 1
  • Priority 2
  • Priority 3
    

IP addresses

• SIP Proxy (IP / GSM) – IP address of the SIP Proxy from which 2N^® VoiceBlue Next awaits the GSM outgoing call requirements. 

• SIP Proxy (GSM / IP) – IP address of the SIP Proxy to which 2N^® VoiceBlue Next turns in the case of a GSM incoming call.
• SIP registrar – IP address of the SIP registration server.

• NAT firewall – IP address for the NAT firewall.
• STUN server – IP address of the STUN (Simple Traversal of UDP through NATs (Network Address Translation)) server for obtaining the public IP address under which 2N^® VoiceBlue Next operates in the Internet network. You are advised to fill in this field if 2N^® VoiceBlue Next operates in a private network separated from the Internet using the NAT or firewall. The default port for sending requests to STUN is 3478.
• Next STUN request (60–6553, 0=off) [s] – used for refreshing information on the public IP address of 2N^® VoiceBlue Next. By editing this item you can configure the frequency of queries sent to the STUN server.

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.

GSM basic parameters

GSM selection

• Assignment of GSM-channel
  • Cyclical
  • Smart - least used minutes
  • Smart - most remaining minutes
  • Linear - always first free module

Count of digits dialled from VoIP

• Minimum digits from VoIP – minimum count of digits to be dialled into GSM.
• Maximum digits from VoIP – maximum count of digits to be dialled into GSM.
• Wait for next digit [s] – timeout during which 2N^® VoiceBlue Next awaits further digits dialled from VoIP to GSM.

Calls

• Relax timeout [s] – time interval between the end of the last call and the beginning of the next call via one and the same GSM module (all incoming and outgoing calls are rejected during this timeout). The recommended value is 2 seconds. Do not change this setting unless absolutely necessary.
• Timeout for ringing to GSM [s] – 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.

Bank holiday list

A list of dates to which the weekend routing mode should be applied in the LCR table. 

DTMF settings

The minimum delay between two identical DTMF characters received [s/100].

Tone detector settings

The GSM gateway can automatically detect user defined tones sent by the GSM/UMTS network during call setup. In general, user defined tones are tones of the number to be transmitted. Having detected such tone, the GSM gateway terminates the call automatically and attempts to set it up through the next available outgoing group (as defined in the LCR table).

• Frequency 1;2;3;4 – defined frequency of the tone to be detected.
• Sequence list – sequence of the above defined tones for detection.

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%.

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.

Wavecom Module Settings

• Transmission volume [dB] – set the module volume for the outgoing direction.

• Reception volume [dB] – set the module volume for the incoming direction
  

• GSM-band selection – select the GSM frequencies for the mobile network.

• Echo cancelling – enable/disable echo cancelling for the module.

• 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.

Telit 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.
• 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.

Audio level DSP

Here you can set the voice level for calls in the signal processor of the GSM gateway.

• Output audio level DSP [dB] – audio volume gain/loss to VoIP
• Input audio level DSP [dB] –  audio volume gain/loss to GSM

Tone generated for incoming calls from GSM/UMTS

• Dialtone – the dialtone type for GSM/UMTS incoming calls.
• Ring tone – the ringing tone type for GSM/UMTS incoming calls.
• Generate busy tone to GSM/UMTS – generation of the busy tone for call end.

Error GSM/UMTS causes

• Here you can set the ISDN release cause for the states mentioned below. The call that meets the below mentioned requirements will be rejected with a user-defined cause (the ISDN cause number will be transformed into a SIP code for VoIP according to the table shown below).
  • Lack of digits in OVERLAP mode – the call will be rejected that fails to meet the minimum digits requirement.
  • Restricted number prefix – the call will be rejected whose prefix has not been found in any of the prefix lists.
  • Selected module / GSM group is not ready – the call will be rejected in case no GSM module is available in the LCR–selected GSM outgoing group.
  • Selected module / GSM groups are not ready – the call will be rejected in case no GSM module is available in the LCR–selected GSM outgoing groups.

Cause translation

Here you can convert the release cause received from GSM/UMTS into another ISDN release cause. The resultant ISN cause number will be transformed into a VoIP SIP code as included in the table below:

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 – fill in this parameter to make the GSM gateway send an SMS message to every called subscriber regardless of whether or not the call was connected. The %N string inserts the CLIP received from VoIP into the SMS.
• Save received SMS to – select the storage for SMS received.
• SIM card identification – select the SIM IMSI/SCID for CDRs.
• Disable CLIP from GSM/UMTS to VoIP – enable/disable resending the CLIP from GSM to VoIP.
• Reject call with CHLD – reject incoming GSM/UMTS calls by means of AT+CHLD (user 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

You can assign the GSM/UMTS modules to groups separately for incoming and outgoing calls. See the two items below for outgoing and incoming group settings.

GSM outgoing groups

2N^® VoiceBlue Next allows you to work with two groups of outgoing calls for each of which you can set variable connection set-up modes and count of used minutes and sent SMS messages for a selected period. 

General settings

• 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.

• 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.
• 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 time between the end of the current call and start of the next call via one and the same GSM/UMTS module. The recommended value for heavy-traffic installations is 2 seconds!

  • Add random time – use this auxiliary parameter to add random time in seconds. Thus, the resultant time is the sum of the two above mentioned parameters.

Network settings

• BTS lock – identify the BTS to which the GSM modules shall log in. Restart the selected GSM modules to execute the changes.

• 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.

Send CLIP from VoIP to GSM/UMTS

• Transfer CLIP to GSM/UMTS – enable/disable the function.
• Separating char – the CDN/CLIP separating character.
• Modify (" removes one digit) – you can change the CLIP. The "– character is used for deleting one char from the left.

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.

• CLIR – fill in this parameter to define whether or not the called subscriber should be displayed the calling SIM card telephone number. It is recommended that the telephone number of the SIM card inserted in the GSM module should not be displayed to the called subscriber to avoid problems with CallBack from GSM.

• Maximum count of called minutes – set the maximum count of minutes to be used via the SIM card per month.
• Count of SMS messages – set the maximum count of SMS messages to be sent via the SIM card per month.
• Day to restore call limit and delete statistics – set the day in a month on which the statistics of the Maximum count of called minutes a Count of SMS messages parameters should be deleted.
• First count – set the length of the first pulse after which the pulse counting will be changed as set in the Next count parameter.
• Next count – set the length of one pulse in seconds after the time as defined in the First count parameter elapses.

• Day limit of called minutes – set the maximum count of minutes to be called via the SIM card per day.

Time limits

Define the SIM card use time limits for the GSM group.
Call tariffs – use this function to assign up to four independent free minute counters to each GSM module (SIM card) group:

• 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 the PSTN
• Tariff 4 = free minutes for calls within a closed user group (VPN)

Set the LCR table (assign prefixes to tariffs) correctly to ensure full functionality of this feature. If routing to a specific tariff is not applied in the LCR table, global free minutes will be used.

• Free minutes – Count of free minutes for a specific tariff offered by the provider.
• Transferred minutes – Maximum count of unused free minutes to be transferred to the next period.
• Day of restoring free minutes – Day on which the free minute counters are to be reset automatically. Choose every 24 hours, a day in a month, or a day in a week for restoring.
• Week of restoring free minutes in month – Define the week in which the free minute counters are to be reset automatically. This setting is useful when counter reset takes place on 'every other Friday in month'. Otherwise, leave the 'every' value. 

GSM incoming groups

2N^® VoiceBlue Next allows you to work with two groups of incoming calls for each of which you can set variable connection set-up modes.

General settings

• Mode – set how the gateway should process incoming GSM calls:
  
  • Reject incoming calls – all incoming GSM calls are automatically rejected.
  • Ignore incoming calls – all incoming GSM calls are ignored. The calling subscriber gets the check ringing tone.
  • Accept incoming calls + voice message – incoming GSM calls are received and, if preset so, the DTMF dial-in with a voice welcome note is activated.
  • Accept incoming calls + dialtone – incoming GSM calls are received and, if preset so, the DTMF dial-in function with a simulated secondary dialtone is activated.
  • CallBack after ring / Reject – in case the CallBack table includes the CLIP, a CallBack is made. CLIP-less incoming calls are rejected.
  • CallBack after ring / Ignore – in case the CallBack table includes the CLIP, a CallBack is made. CLIP-less incoming calls are ignored.
  • Report to PC + voice message – the GSM gateway sends information on incoming calls to a PC equipped with the call routing software. The DTMF dial-in with a voice welcome note can be activated for incoming calls.
  • Report to PC + dialtone – the GSM gateway sends information on incoming calls to a PC equipped with the call routing software. The DTMF dial-in with a simulated secondary dialtone can be activated for incoming calls.
• Minimum digits in DTMF – set the minimum count of digits to be required by the gateway for DTMF dial-in.
• Maximum digits in DTMF – set the minimum count of digits to be accepted by the gateway for DTMF dial-in.
• DTMF dialling timeout [s] – set the time for the GSM gateway to wait for the first / next DTMF digits. If you set this parameter to 0, the incoming call is automatically connected to the numbers included in the List of called numbers.
• Day of deleting GSM incl. group statistics – define a day in a month on which the statistics on incoming calls should be deleted.
• Prefix before DISA dial-in – enter a numerical prefix to be automatically inserted before the DTMF dial-in.
• CLIP – use this parameter to modify the CLIP from GSM/UMTS networks. '+' is removed automatically from international identifiers. Use '–' to remove a digit. Examples (GSM CLIP +420600123456):

• Looping of voice message – set the voice message playing time.

Send CLIP from GSM/UMTS to VoIP

• Transfer CLIP from GSM/UMTS – enable/disable the function.
• Separating char – enter the character to separate the calling SIM card CLIP and ID extension to be called.
• Modify – modify the ID extension.

Others

• Time to keep CLIP in table – set the time for which records are to be kept in the AutoCLIP routing table.
• Add record only for unconnected call – unconnected outdoing calls are stored in the AutoCLIP table only.
• Delete record for connected answer – if the CallBack is successful, the corresponding AutoCLIP record will be deleted.
• Skip DTMF for numbers not in CLIP Routing table – enable this function to define that only the incoming calls whose CLIPs are stored in the CLIP Routing table will be able to use DTMF dial-in. 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 define that the incoming call will not be forwarded to the number(s) included in the List of called numbers in case the call is rejected after the preceding DMTF dial-in.
• Auto end to CallBack request – any incoming call whose CLIP matches the CallBack setting will be rejected and all the other incoming calls will be ignored. In either case, the CallBack function will be invoked after the incoming call ends.

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

The window helps you adapt the gateway to calling to variable GSM networks. You can set the prefix-based call routing rules here and identify up to sixteen different groups for routing.

Prefix list 1–16

Sixteen prefix groups to be assigned in the LCR table. 

Basic settings

• GSM network ID – name the prefix list for convenience.
• Default count of digits – the default length of the number to be dialled into GSM/UMTS when routing via the selected prefix list. Use this parameter if the Digits count parameter is not included in the Accepted prefixes table. 

Table of replaced prefixes

This table helps you change the prefix of the number received (from 00 to +, e.g.). You can just add or remove a prefix using this table. This change is performed before the prefix table search is initiated. Keep the „"/ „" record at least in the table for a proper function. 

Table of accepted prefixes

A list of prefixes of called destinations to which the selected prefix list applies.

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 VoIP interface.

• From (channels/groups) – source channels or groups via which calls are routed to 2N^® VoiceBlue Next.
  • GSM ALL – any of the GSM incoming groups can be used for call routing.
  • GSM GRP1–4 – define one GSM incoming group or a range of GSM incoming groups 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 16 prefix lists.
  • Prefixlist 1–16 – call routing will obey the rules set in Prefix list 1–16. The count of digits to be dialled is governed by the Prefixes setting. 
  • All prefixes – all the prefix lists are permitted. Prefix lists 1–16 will be searched from 1 to 16. 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 – 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^® VoiceBlue Next. If the defined interface is inactive or the outgoing GSM group tariff is exhausted, the next row will be applied.
  • GSM groups 1–4 – outgoing GSM groups. Set the tariff to be used. Refer to the GSM outgoing groups 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. 

CLIP routing table + CallBack

The table permanently assigns CLIPs to the extensions to which incoming calls are to be routed automatically and set the CLIP list for the GMS network for which the CallBack function 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 – VoIP 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).

Mobility Extension

The table registers the Mobility Extension users.

• Name – enter the user name for calling to VoIP.
• User – enter the registration user name for VoIP.
• Password – enter the registration user password for VoIP.
• GSM number (CLIP) – enter the user SIM card CLIP.
• Follow me function – enable/disable call forwarding to a mobile user station (based on CLIP).
• SMS at no answer function – enable/disable sending of information SMS on missed calls.

Ethernet configuration

This window helps you configure the Ethernet interface of your gateway.

• Use DHCP – enable/disable the DHCP client function in 2N^® VoiceBlue Next.
• IP address – the fixed IP address (v4) of the 2N^® VoiceBlue Next Ethernet interface.
• Subnet mask – the subnet mask for the 2N^® VoiceBlue Next Ethernet interface.
• Default gateway – The 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. 

Login configuration

The window helps you set the access password and name for the 2N^® VoiceBlue Next web interface. Use the same data for Telnet connection too.

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 – here you can change the login window graphic lay-out. 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.