en

#auth
AUTHORIZE="Autenticazione"
AUTH_NAME="Nome Utente"
AUTH_PASSSWD="Password"
AUTH_LOGIN="Login"

#system
SYS_LANG="Lingua"
SYS_BUTTON_DEFAULT="Ripristina"
SYS_BUTTON_SAVE="Applica"
SYS_BUTTON_SAVE_AND_RESTART="Applica e riavvia"
SYS_BUTTON_BROWSE="Sfoglia"
SYS_NOSCRIPT1="JavaScript non abilitato."
SYS_NOSCRIPT2="La pagina non verrà caricata correttamente"
INVVAL_INT_MINMAX="Valore non valido. Inserire un valore numerico"
INVVAL_LENGTH_MINMAX="Valore non valido. Inserire un valore alfabetico"
INVVAL_IP="Valore non valido. Inserire un indirizzo IP valido"
INVVAL_ENTRY="Valore non valido"

#menu
MENU_NET="Parametri di rete"
MENU_AUDIO="Codec audio"
MENU_VIDEO="Codec video"
MENU_VRATNY="Configurazione semplificata"
MENU_TRIGGERS="Relays"
MENU_TIMES="Impostazione orologio"
MENU_NUMB="Programmazione tasti"
MENU_GSMBR="Parametri GSM"
MENU_GSMX="Modulo GSM"
MENU_GSMXDIR="Chiamate permesse"
MENU_GSMNUM="Modifica selezione"
MENU_SIP="Parametri SIP client"
MENU_SIPSRV="Parametri SIP server"
MENU_ADMIN="Manutenzione"
MENU_GOTO_VIDEO="Video"
MENU_HELP="Help"
MENU_STAT_DAY="Servizio giorno"
MENU_STAT_NIGHT="Servizio notte"
MENU_SET_LANG="Configura"

#network
NET_TIT="Parametri di rete"
NET_HOSTNAME_TIT="Hostname"
NET_DHCP_TIT="Configurazione via DHCP"
NET_DHCP_ID_TIT="ID client DHCP"
NET_IP_ADDR_TIT="Indirizzo IP"
NET_NET_MASK_TIT="Maschera di rete"
NET_NET_GATEWAY_TIT="Default gateway"
NET_DNS_SERVER1_TIT="Server DNS primario"
NET_DNS_SERVER2_TIT="Server DNS secondario"

#audio
AUD_PRIORITY="Priorita'"

#video
VID_IMAGE_SIZE_TIT="Dimensione"
VID_IMAGE_PER_SEC_TIT="Immagini per sec."
VID_BRIGHTNESS_TIT="Luminosita'"
VID_COLOUR_TIT="Colore"
VID_CONTRAST_TIT="Contrasto"
VID_HUE_TIT="Tinta"
VID_AUTOBRIGHT_TIT="Luminosita' auto"
VID_QUALITY_TIT="Qualita'"
VID_WHITENESS_TIT="Bilanciamento"
VID_LIGHT_FREQ_TIT="Frequenza"
VID_OPEN_ERR="La telecamera non è corretta o non funziona!"

#spinace
TRIG1="Relay 1"
TRIG2="Relay 2"
TRIG_MOD="Modalita'"
TRIG_EXT_DAYNIGHT="Codice esterno giorno + notte"
TRIG_EXT_DAY="Codice giorno esterno"
TRIG_EXT_NIGHT="Codice notte esterno"
TRIG_IN_PHONE="Codice interno da tel."
TRIG_TIME_RUN="Tempo chiusura relay [sec]"
TRIG_HANDLE_INCALL="Controllo chiam. entr."
TRIG_TIME_DELAY="Ritardo tra 1 e 2 in mod. 5"

#casove parametry
TIM_TIME_CALL="Max durata conversazione [min]"
TIM_NUMBER_RING="N. di squilli"
TIM_BETW_PUT_BUTT="Timeout di digitazione [sec]"
TIM_TIME_CALLDOWN="Timeout di selez. autom.[sec]"
TIM_TIME_BEFORE_CALL="Tempo di selezione[sec]"
TIM_RING_BEGIN_END="Segnalaz. audio-apri/chiudi"
TIM_OTHERS_RINGS="Segnalaz. audio- altri toni"

#vratny - zakladni nastaveni
BASE_MODE_NUMBSELECT="Modalita' scelta numeri"
BASE_MODE_NUMBSELECT_GR_NUMB="Mod. 2 gruppi"
BASE_MODE_NUMBSELECT_DN="Giorno-Notte"

BASE_CHAR_CALL="Prolung. conversazione"
BASE_CHAR_CALL_HV="* - asterisco"
BASE_CHAR_CALL_KR="# - cancelletto"

BASE_CHAR_CALLDOWN="Codici relays"
BASE_CHAR_CALLDOWN1="Cod. apertura relay 1"
BASE_CHAR_CALLDOWN2="Cod. apertura relay 2"

BASE_SWITCH="Passaggio Giorno/Notte"
BASE_SWITCH_MANUAL="Manuale"
BASE_SWITCH_AUTO="Automatico"

BASE_CODE="Codice di commutazione"
BASE_CODE_NIGHT="Codice per mod. Giorno"
BASE_CODE_DAY="Codice per mod. Notte"

BASE_KEYB_ONPOSITION="Posizione tastiera"

BASE_KEYB_MODE="Mod. tastiera"
BASE_KEYB_MODE_DIRECT="Selez. diretta (telefono)"
BASE_KEYB_MODE_MEMORY="Selez. numeri brevi"

GUARD_LOAD_FAIL="Caricamento fallito"
GUARD_SAVE_FAIL="Salvataggio fallito"
GUARDSTAT_ERR="Errore. Lo status non è stato caricato completamente."

#pametova tlacitka
MEM_BUTT="Tasto"
MEM_NIGHT="Gruppo NOTTE"
MEM_DAY="Gruppo GIORNO"

#gsm gate
GSMBR_TIT="Parametri GSM"
GSMBR_DISA_TIT="DISA"
GSMBR_OGM_TIT="OGM"
GSMBR_DIRECT_TIT="Accesso diretto"
GSMBR_REC_TIT="rec"
GSMBR_OPWAIT_TIT="Attesa"
GSMBR_ERSCLP_TIT="Cancella CLIP"
GSMBR_AEC_TIT="Cancella eco"
GSMBR_LOAD_FAIL="Errore nel caricamento dei parametri"
GSMBR_SAVE_FAIL="Errore nel salvataggio dei parametri"
GSMBR_DEFAB_TIT="Precedenza al modulo 1"
GSMBR_0BCLP_TIT="Anteponi 0 al CLIP"

#gsm brana - kanal
GSMX_TIT="Modulo GSM"
GSMX_PIN_TIT="PIN"
GSMX_MSN_TIT="N. destinatario"
GSMX_CALLCHARG_TIT="Tariffazione"
GSMX_PULSE_EACH_TIT="durata scatto"
GSMX_VOL_GSM_TIT="Volume GSM"
GSMX_VOL_ISDN_TIT="Volume VoIP"
GSMX_CALLIN_TIT="Abilita chiamate entranti"
GSMX_CALLOUT_TIT="Abilita chiamate uscenti"
GSMX_PROGRESS_TIT="Tono controllo chiamata"
GSMX_REDIR_TIT="trabocco su GSM "
GSMX_SMCALLBACK_TIT="Richiamata intelligente"
GSMX_0_TIT="0"
GSMX_CLIR_TIT="CLIR"
GSMX_ROAMING_TIT="Roaming"

#gsm brana - povolene smery
GSMXDIR_TIT="Modulo direzioni consentite"
GSMXDIR_FROM_TIT="da"
GSMXDIR_TO_TIT="a"

#gsm brana - prirazeni cisel
GSMNUM_TIT="Tabella numeri interni"
GSMNUM_NUM="Numero"
GSMNUM_ADDR="Indirizzo IP"

#gsm brana - status
MENU_GSMSTAT="Stato del Gateway GSM"
GSMSTAT_TIT="Stato del Gateway GSM"
GSMSTAT_MODULE_TIT="Modulo"
GSMSTAT_CHANNEL_TIT="Frequenza canale"
GSMSTAT_STRENGTH_TIT="Potenza segnale"
GSMSTAT_BCCH_TIT="Potenza BCCH"
GSMSTAT_COUNTRY_TIT="prefisso Paese"
GSMSTAT_NETCODE_TIT="Prefisso rete"
GSMSTAT_AREA_TIT="Codice di zona"
GSMSTAT_CELLID_TIT="ID cella"
GSMSTAT_IMSI_TIT="IMSI"
GSMSTAT_SWVERSION_TIT="Versione firmware GSM"
GSMSTAT_ERR="Errore nel caricamento dei parametri."

#sip parametry
SIP_TIT="parametri SIP client"
SIP_SERVER_TIT="Server SIP"
SIP_SERVER_IP_TIT="Indirizzo IP"
SIP_SERVER_PORT_TIT="Porta"
SIP_ACCOUNT_TIT="Account"
SIP_ACCOUNT_MOD_TIT="Modulo account"
SIP_ACCOUNT_NAME_TIT="Nome utente"
SIP_ACCOUNT_PASS_TIT="Password"
SIP_ACCOUNT_EXPIRE_TIT="Scadenza [sec]"
SIP_REG_SUCCESS="Registrazione avvenuta"
SIP_REG_FAIL="Registrazione fallita"

#services
ADMIN_TIT="Menu Manutenzione"
UPLOAD_VERSION="Versione FW VoIP"
UPLOAD_GUARD_VERSION="Versione FW KIRO"
DOWNLOAD_LOGFILE="Scarica il file di log"
ADMIN_SHOW_LOGCALL="Mostra log chiamate"
ADMIN_SHOW_LOGREG="Mostra il registro dei log"
ADMIN_SHOW_LOGVOIP="Mostra il log VoIP"
ADMIN_ENABLE_LOG="log esteso"
ADMIN_DISABLE_LOG="log base"
ADMIN_NTP_SERVER="Time server"
ADMIN_SYSLOG_SERVER="Syslog server"
ADMIN_FW_FILE="Aggiorna firmware"
ADMIN_LANG_FILE="Aggiorna lingua"
ADMIN_CONF_SAVE="Salva configurazione"
ADMIN_CONF_LOAD="Aggiorna configurazione"
ADMIN_FW_SUBMIT="salva"
ADMIN_RESTART="riavvia"
ADMIN_WAIT_RESTART_TIT="Riavvio in corso ..."
ADMIN_PASSWD_TIT="Password di manutenzione"
ADMIN_PASSWD2_TIT="Ripetere password"
ADMIN_PASSWD_DIFFER="Password errata!"

#sip server
SIPSRV_TIT="Parametri Server SIP"
SIPSRV_CLR_SIP_SETTING="Prima di abilitare il server SIP interno si deve disab il server SIP esterno nel menu Parametri SIP."
SIPSRV_ENABLE_TIT="Abilita server SIP"
SIPSRV_REALM_TIT="Nome server(realm)"
SIPSRV_PREFIX_TIT="Prefisso"
SIPSRV_USER_TIT="Numero"
SIPSRV_PASS_TIT="Password"
SIPSRV_USER_CONN="Connesso"

#den noc
MENU_DAYNIGHT="Intervalli giorno"
DAYNIGHT_INTERVAL_TIT="Intervallo"
DAYNIGHT_COMMENT="La riga vuota e' considerata come giorno intero.<br>Le ore non incluse negli intervalli sono consid. NOTTE"
WEEK_START="DOM"
DOW_MON="Lun"
DOW_TUE="Ma"
DOW_WED="Mer"
DOW_THU="Giov"
DOW_FRI="Ven"
DOW_SAT="Sab"
DOW_SUN="Dom"

#curr time
CURRTIME_NOTSET="Ora non configurata"
CURRTIME_NTP_NOTSET="Time server non configurato"

#config load
LOAD_CONF_TIT="Caricamento in corso..."
LOAD_CONF_SUCCESS="Configurazione caricata con successo"
LOAD_CONF_ERROR="Errore durante il caricamento dati"

#user interface
MENU_UI="Interfaccia Utente"
UI_VID_WWW_ENABLE="Video su pag. iniziale"
UI_VID_RTP_ENABLE="Video in chiamate VoIP"
UI_HTTP_PORT="Porta tcp  dell'interfaccia Web"
UI_TELNET_ENABLE="Abilita telnet"


#ENDOFSTRINGS
if [ -z "$MKHELP" ]; then return; fi


HELP_NET_GUARD="
<ul>
<li>Indirizzo IP è quello usato per la connessione al citofono IP se il server SIP interno non viene usato
<li>L'indirizzo Ip del server SIP viene usato per il server SIP interno.
    Se il server SIP interno viene abilitato, tutti i client si connettono esclusivamente tramite questo indirizzo
<li>Il gateway di rete e i server DNS servono solo se ci si connette o registra via Internet.
    Se il citofono IP viene usato solo su rete locale, lasciare vuoto.
<li>Il gateway di rete generalmente non è l'indirizzo del server SIP interno o esterno.
</ul>
"

HELP_NET_GSM="
<ul>
<li>Indirizzo IP 1 e 2 sono indirizzi di Gateway GSM.<br>
    In modalita' P2P si chiama tramite questi indirizzi<br>
    Se si usa un server SIP interno o esterno, si chiama tramite l'indirizzo IP del server SIP
<li>L'indirizzo Ip del server SIP viene usato per il server SIP interno,
    se abilitato, tutti i client (telefoni) chiamano tramite questo indirizzo IP
<li>Il gateway di rete e i server DNS servono solo se ci si connette o registra via internet.
    Se il gateway GSM viene usato solo su rete locale, lasciare vuoto.
<li>Il gateway di rete generalmente non è l'indirizzo del server SIP interno o esterno.
</ul>
"

HELP_AUDIO="
<ul>
<li>in caso di scarsa qualita' audio, provare una diversa priorita' dei codec
<li>nel menu 'Codec audio' scegliere G711&micro; come primo, G711a per secondo, ...
<li>controllare la configurazione audio del client (telefono IP). Almeno G711&micro; o G711a devono essere abilitati
</ul>

"
HELP_VIDEO="
<ul>
<li>In caso di rete lenta, provare a impostare il n. di frame al secondo su un valore piu' basso
    e/o ridurre il formato dell'immagine
</ul>
"

HELP_GSMBR="
<ul>
<li>DISA, se nel codice di selezione diretta viene config un certo numero di cifre, il numero sara' maggiore di
    zero, chi chiama potra' digitare direttam il n. di interno desiderato usando la selezione a toni.    
<li>Attesa,n. di secondi che l'operatore automatico attende l'inserim di un num. di interno.
<li>Cancella Clip	è il parametro per eliminare le prime cifre iniziali di un num. entrante. 
<li>Moduli diretti. Il Gate non usa LCR. L'indirizzo IP 1 viene assegnato in automatico alla SIM nel modulo 1 
    e l'indirizzo IP 2 alla SIM nel modulo 2.
</ul>
"

HELP_GSMX="
<ul>
<li>Codice PIN della SIM inserito nel modulo GSM.
<li>Il numero telefonico viene scelto per salvare la selez autom nell'operatore.
<li>Volume GSM,il volume ISDN permette di regolare il volume delle chiamate in entrambe le direzioni.
<li>Chiamate entranti, chiamate uscenti è l'abilitazione delle chiamate in ingresso/uscita.
<li>Call progress tones attiva/disattiva i toni di inoltro chiamata sulla rete GSM.
<li>Redirection to GSM 2 allow 'overflow' of the enabled directions to the GSM modul 
<li>Smart callback store all outgoing calls which has been missed or refused. When called part calling back
    then the call is automatically routed to extension which made the call.
<li>0, CLIR. It adds 0 or code for Switch OFF outgoing CLIP before each outgoing dialled number.
</ul>
"

HELP_GSMXDIR="
<ul>
<li>Permited calls. If the table is not filled in, all calls are enabled. By filling in the tables of both modules,
    it is possible to achieve automatic re-routing of the outgoing call in the module allowing cheaper calling
    through the given operator (LCR). 
</ul>
"

HELP_GSMSTAT="
<ul>
<li>BCCH strenght is signal strength. -113 to -99 dBm is very bad signal, -98 to -83 dBm is bad signal,
    -82 to -71 dBm is good signal and -70 to -51 dBm is very good signal.
<li>GSM Firmware version is SW version for GSM part.
</ul>
"

HELP_VRATNY="
<ul>
<li>Mode of DoorPhone choice selects number per Day/Night DoorPhone mode or selects numbers of the first and 
    second groups. 
<li>Sign for call extension * or # (10sec before call end the DoorPhone will send a notice, then the call 
    may be extended) 
<li>Two commands in order to hang up the DoorPhone using both switches [2 digits].The advantage is to set 
    the same command both for switch closing and command to guard hanging up. 
<li>Command for DAY / NIGHT mode switching
    <i>Note: The switchover to Day/Night mode remains set in guard even after power supply disconnection.</i>
<li>Dialing as on normal telephone (all number of called person should be pressed on keyboard) 
      - recommended for use SIP proxy server. 
    Only 2-digit memory number is entered on keyboard by 
      which the number of called person is stored (memory number corresponds to button number with respect 
      to Day/Night switchover) - recommended for use P2P.
<li>Connect keyboard or NC-mod4<br>
        =0 only NC-mod4 connected to the basic module <br>
        =1 the keyboard connected on the first position (after IPNCx-mod) <br>
        =2 the keyboard connected on the second position (after first NC-mod4) <br>
        =3 the keyboard connected on the third position (after second NC-mod4) <br>
    <i>Note:The keyboard module is only connected by flat cable as well as NC-mod4 module. The only difference 
    is that keyboard module is always the last in row (no other module can be connected behind it). Connect 
    on the first place (to output of the IPNCx-mod) or the second (to output of the first NC-mod4) positions 
    or third place. It means that 4 to 24 buttons with direct dialing can be used instead of keyboard (per assembly). 
    Pay attention when programming - the position of keyboard connected must be correctly specified.
    The choice is entered by gradual pressing of buttons with digits. Firstly the key symbol must be pressed to enter 
    a password. When pressing X, the DoorPhone will hang up or cancel your dial. Button with key symbol use for 
    'Point' in IP adress in P2P mode.</i>
</ul>
"

HELP_TRIGGERS="
<ul>
<li>Relay mode:<br>
     =1 switch mode - it will close on command or password for period t1/2 (used for electrical locks, gate opening etc.)<br>
     =2	camera mode - it will close by guard pick up and open by hanging up.<br>
     =3	lighting mode - it will close by guard pick up and stay closed even for period t1/2 after guard hanging up. <br>
     =4 bell mode - it will close after button pressing and open after period t1/2 (used for e.g. external bell or 
        horn connections).<br>
     =5 gradual opening mode - in this mode the only relay 2 will be set together with relay 1 set to mode 1. 
        The relay 1 is activated for period t1, then the time t3 is proceeding before relay 2 closing. Then the 
        relay 2 is activated for t2 period and afterwards the DoorPhone hangs up.  <br>
	<i>Note: The only relay 1 can be activated from phone and all sequence started. Besides that the relay 2 can
        be separately activated from buttons by password.</i>
<li>password for relay closing from buttons or keyboard [2 to 6 digits]. Total 6 passwords, they are controlled 
    by Day/Night; the combination is entered either by DoorPhone buttons (first 10 buttons) or from attached
    (after pressing of key symbol). The relay closing influences the set switch mode and  Day/Night  
    By setting of choice mode of 2 number groups the DoorPhone is permanently in DAY mode.
    By password choice some rules have to be observed:<br>
     - Select passwords in way not to find its combination out from wear of certain buttons by frequent use.<br>
     - Select the first password button from frequentless button for direct dialing (-extends choice time)
       (-not valid for keyboard).<br>
     - Pay attention to congruity of password numbers when one password includes other one, e.g. relay 1 has 1234 
       and relay 2 has 12345. Then after pressing button 4 the only relay 1 is called, but password choice 234 for 
       relay 2 can call both relays after pressing switch 4.
<li>Command from phone after relay closing [2 digits]. The same command can be set for both relays, 
    then they are activated at the same time. The advantage is to set the same command both for relay closing 
    and command to DoorPhone hanging up. 
<li>Duration of relay closing in second [2 digits 01-99] 
<li>To prohibit the control during incoming call is important e.g. when using relay 2 in mode 1 for control of 
    garage gate opening, when the electronics opens the gate and the gate is closed by car passage. Then the control 
    from phone could undesirably cause the permanent gate opening (not closed - no car passage).
<li>time in second between close relays 1 and 2 by mode setting of relay 2 is 5 (gradual opening) [2 digits 01-99] 
</ul>
"

HELP_TIMES="
<ul>
<li>max. time, for which the DoorPhone is hanging up, this time can be extended during call by sign choice from 
    telephone (* or #)
<li>Number of incoming call rings, the DoorPhone pick up after preseted number of rings. After detection first 
    ring - LED on front panel blinking. The number can be set from 1 to 9.
<li>max. time [sec] among button presses [range 1-9]
    normal buttons<br>
     - switch closing - if time between two next presses is bigger than w  time, the code is not evaluated correctly.<br>
     - dialing - if the button, we are pressing, is the first password number for switch closing, so the choice is 
       delayed by this w time.<br>
    keyboard<br>
     - switch closing - if time between two next presses is bigger than w  time, the code is not evaluated correctly.	<br>
     - dialing the same as of phone, if time after the last pressed button is bigger than w time, then the dialing starts.
       If the number is incomplete, it is necessary to hang up (X button) and the dialing will be repeated.<br>
     - dialing from memory, if time following the first pressed button is longer than w time, then the entry of memory 
       number has to be repeated.
<li>time [sec] for which the guard will hang up, before repeated dialing (button pressing during call or dialing, 
    busy tone detection) [range 1-5]
<li>after finishing the dialing it calculates time (ringing tones).  If the number exceeds time in second, it will 
    hang up [range 10-99]. The dialing is repeated in case, when the dialing mode of 2 groups is set.
<li>In default is status of DoorPhone signalling acoustically. If signalling makes problem, so this signalling 
    pick up / hang up prohibited. 
<li>In default is status of DoorPhone signalling acoustically. If signalling makes problem, so this signalling 
    others tones prohibited.
</ul>
"

HELP_NUMB="
<ul>
<li>telephone number up to 16 digits, we want to store. The numbers are the numbers of the first group or numbers 
    of Day mode. In default setting is table memoirs empty. While using setting P2P to the memoirs saves IP address  
    e.g . 192*168*1*250, where '*' means '.' , while using SIP proxy server to the memoirs saves phone number e.g. 117.
<li>telephone number up to 16 digits, we want to store. The numbers are the numbers of the second group or numbers 
    of Night mode. In default setting is table memoirs empty. While using setting P2P to the memoirs saves IP address  
    e.g . 192*168*1*250, where '*' means '.' , while using SIP proxy server to the memoirs saves phone number e.g . 117.
   <i>Note: The switchover to Day/Night mode remains set in DoorPhone even after power supply disconnection.</i>
</ul>
"

HELP_SIP="
<ul>
<li>SIP proxy server - here enter IP address and port (if differ from default 5060) of server to send registration to
    or route calls
<li>name and password are not mandatory but must be set axactly same as on SIP server machine
<li>after entered data saved, registration attempt is executed (if name is not empty) and result shown
<li>if the registration is not successfull, you will see reason in Registration log in menu Services.
</ul>
"

HELP_SIPSRV="
<ul>
<li>The Realm is name send from SIP server to client as the SIP server name at make registration or call.
<li>Prefix 1 and 2 are 1 to 3 digit numbers which is prepended before called GSM number from IP phone.<br>
    If you prepend prefix 1 your call will be routed via GSM module 1,
    if you prepend prefix 2 your call will be routed via module 2.<br>
    Both prefixes must not be the same and if leave empty, call via appropriate GSM Module will not be possible.
<li>Client numbers and passwords - must be entered the same as in client settings,
    in other case client will not be able to register itself on internal SIP server and make call
</ul>
"

HELP_ADMIN="
<ul>
<li>Download log file. If you are in troubles and need technical support, you will need this file. Follow these steps:
  <ol>
  <li>press 'enhanced log' button
  <li>do the problematic action, this action will be recorded step by step into log file
  <li>click on 'Download log file', save the file and send it to technical support
  </ol>
<li>Show call log<br>
    - record of few last incomming and outgoing calls. In case of error, reason is also shown.
<li>Show registration log<br>
    - registration on SIP server, in case of error reason is shown<br>
    - successful registration is always done in two steps. Firstly a client sends the request to server and
      server responds with its realm, in second step the client sends identity based on realm of server and
      server responds with success or access denied
<li>Syslog server is computer able of receiving internal messages from device
<li>In Firmware upgrade customer is allowed to load new versions of firmware into device.
  Here you also load customizations.<br>
  If upgrading firmware is in progress do not power off device, you risk the device get unusable!<br>
  After firmware upgrade pressing reset button is allways required.
<li>Upload language - possibility to supply custom language.
  Name of this file appears in Language combo box. HTML entities in file name are allowed.
  Language file must start with line cs or en - the language from which thist customization is derived.
<li>Service password, change the manufacturer default password is highly recommended.
</ul>
"

TROUBLESHOOTING="
<h2>Troubleshooting</h2>
Many of problems is possible successfully solve by cooperation with technical support.
In this case tech. support needs exact and clean description of your problem and log file downloadable from menu Support.
Follow these steps:
<ol>
<li>press 'enhanced log' button
<li>do the problematic action, this action will be recorded step by step into log file
<li>click on 'Download log file', save the file and send it to technical support with description of your problem
</ol>

<h3>Registration</h3>
<ul>
<li>registration is unsuccessful<br>
    - go to menu Service and click 'Show registration log', record of registration attempts and results will appear

<li>registration is unsuccessful - in registration log is reason: Timeout<br>
    - SIP server is unreachable, check SIP server address setting in menu 'SIP parameters'.<br>
    - check network connection and SIP server running

<li>registration is unsuccessful - in registration log is reason: 404 (Not found)<br>
    - check IP address of SIP server, port and registration name in menu 'SIP parameters'

<li>registration is unsuccessful - in registration log is reason: 'Unauthorized' or 'Access denied'<br>
    - registration is allways done in two steps, in first step client obtain server's realm and result is 'Unauthorized',
      in secont step client sends authorisation and should be successful. Two step registration is OK.<br>
    - registration name (number) and password are not mandatory, but must be set exactly same as on SIP server<br>
    - look into SIP server machine log, you may find interesting things

<li>registration is unsuccessful - nothing helps<br>
    - in menu Service download log file and send it with problem description to technical support
</ul>

<h3>Call</h3>
<ul>
<li>connection of call is not possible<br>
    - in menu Service click on 'Show call log', window with call record and possible errors will display
<li>called number in call log is not desired GSM number<br>
    - called number must be same as desired GSM number, if it differs, your SIP server is misconfigured
<li>in call log I see 'Bypass SIP server'<br>
    - the calling client has not properly setup SIP server.<br>
    - fix client setting, it must call via SIP server, not GSM Gate IP address directly<br>
    - try set IP address instead of host (domain) name for SIP server in menu 'SIP Setting'
<li>in call log I see 'Unsupported media type'<br>
    - in 'Setting audio' choose as priority one codec G711&micro;, second G711a and so on<br>
    - check client setting (phone), must have codecs G711&micro; or G711a enabled
<li>call is not possible, or is early lost, nothing helps<br>
    - in menu Service download log file and send it with description to technical support
</ul>

<h3>Audio</h3>
<ul>
<li>poor quality<br>
    - in 'Setting audio' choose as priority one codec G711&micro;, second G711a and so on<br>
    - check client setting (phone), must has codecs G711&micro; or G711a enabled
<li>poor quality persists<br>
    - try other codec combinations
<li>audio setting is OK, but still poor quality<br>
    - you may try record network traffic of call by analyzer named wireshark (download from www.wireshark.org)<br>
    - this record with description and log file from menu Service send to technical support
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