en

#auth
AUTHORIZE="Authorize Page"
AUTH_NAME="Login name"
AUTH_PASSSWD="Password"
AUTH_LOGIN="Login"

#system
SYS_LANG="Language"
SYS_BUTTON_DEFAULT="default values"
SYS_BUTTON_SAVE="save changes"
SYS_BUTTON_SAVE_AND_RESTART="save and restart"
SYS_BUTTON_BROWSE="browse"
SYS_NOSCRIPT1="JavaScript is not enabled."
SYS_NOSCRIPT2="This pages will not work properly !!!"
INVVAL_INT_MINMAX="Invalid entry. Enter number in range"
INVVAL_LENGTH_MINMAX="Invalid entry. Required chars"
INVVAL_IP="Invalid entry. Enter valid IP address"
INVVAL_ENTRY="Invalid entry"
FLASH_SAVE_FAIL="Save your settings in memory failed."
FLASH_LOCK_FAIL="Another connection changed data. Save failed."

#menu
MENU_NET="Network setting "
MENU_AUDIO="Setting audio"
MENU_VIDEO="Setting video"
MENU_VRATNY="Basic parameters"
MENU_TRIGGERS="Relays"
MENU_TIMES="Time parameters"
MENU_NUMB="Memory numbers"
MENU_GSMBR="GSM gate"
MENU_GSMX="GSM module"
MENU_GSMXDIR="Accepted calls"
MENU_GSMNUM="Number translation"
MENU_SIP="SIP parameters"
MENU_SIPSRV="SIP server"
MENU_ADMIN="Service"
MENU_GOTO_VIDEO="Video"
MENU_HELP="Help"
MENU_STAT_DAY="Day mode"
MENU_STAT_NIGHT="Night mode"
MENU_SET_LANG="Set"

#gsm mode
GSM_MODE_TIT="Select mode"
GSM_MODE_1x2_TIT="Two-channels GSM gateway"
GSM_MODE_2_2_TIT="Two independent GSM gateways"
GSM_MODE_1_1_TIT="Single-channel GSM gateway"

#network
NET_TIT="Network setting"
NET_HOSTNAME_TIT="Display name"
NET_DHCP_TIT="Setup via DHCP"
NET_DHCP_ID_TIT="DHCP client ID"
NET_IP_ADDR_TIT="IP address"
NET_NET_MASK_TIT="Network mask"
NET_NET_GATEWAY_TIT="Default gateway"
NET_DNS_SERVER1_TIT="Primary DNS server"
NET_DNS_SERVER2_TIT="Secondary DNS server"
NET_NAT_ADDR_TIT="NAT address"

#audio
AUD_PRIORITY="Priority"
AUD_AEC_TITLE="Acoustic Echo Limiter"
AUD_AEC_THRESHOLD="Threshold"
AUD_AEC_DIVIDER="Divider"
AUD_AEC_SAMPLES="Length"

#video
VID_IMAGE_SIZE_TIT="Image size"
VID_IMAGE_PER_SEC_TIT="Numbers image per sec."
VID_BRIGHTNESS_TIT="Brightness"
VID_COLOUR_TIT="Colour"
VID_CONTRAST_TIT="Contrast"
VID_HUE_TIT="Hue"
VID_AUTOBRIGHT_TIT="Auto brightness"
VID_QUALITY_TIT="Quality"
VID_WHITENESS_TIT="Whiteness"
VID_LIGHT_FREQ_TIT="Light frequency"
VID_PRIORITY="Video codec priority"
VID_MCAST_ADDR_TIT="Video multicast address"
VID_OPEN_ERR="Camera is not connected, or doesn't operate!"

#spinace
TRIG1="Relay 1"
TRIG2="Relay 2"
TRIG_MOD="Relay mode"
TRIG_EXT_CODE="External code"
TRIG_EXT_DAYNIGHT="External code day + night"
TRIG_EXT_DAY="External code day"
TRIG_EXT_NIGHT="External code night"
TRIG_IN_PHONE="Internal code from phone"
TRIG_TIME_RUN="Relay closing [sec]"
TRIG_HANDLE_INCALL="Control of incoming call"
TRIG_EXT_COND_TIT="External code enable"
TRIG_EXT_COND_ALWAYS="always"
TRIG_EXT_COND_NEVER="never"
TRIG_EXT_COND_ETHER="if net is down"
TRIG_TIME_DELAY="Delay between 1 and 2 in mode 5"

#casove parametry
TIM_TIME_CALL="Maximum call duration [min]"
TIM_NUMBER_RING="Numbers of rings"
TIM_BETW_PUT_BUTT="Time between key presses [sec]"
TIM_TIME_CALLDOWN="Time hang up before redial [sec]"
TIM_TIME_BEFORE_CALL="Time before redial [sec]"
TIM_RING_BEGIN_END="Audio signaling - opening/closing"
TIM_OTHERS_RINGS="Audio signaling - others tones"

#vratny - zakladni nastaveni
BASE_MODE_NUMBSELECT="Mode of choice numbers"
BASE_MODE_NUMBSELECT_GR_NUMB="2 group of numbers"
BASE_MODE_NUMBSELECT_DN="Day-Night"

BASE_CHAR_CALL="Prolongation char"
BASE_CHAR_CALL_HV="* - star"
BASE_CHAR_CALL_KR="# - hash"

BASE_CHAR_CALLDOWN="Hang up phone"
BASE_CHAR_CALLDOWN1="Code for hang up phone 1"
BASE_CHAR_CALLDOWN2="Code for hang up phone 2"

BASE_SWITCH="Switching between Day Night"
BASE_SWITCH_MANUAL="Manually"
BASE_SWITCH_AUTO="Automatic"

BASE_CODE="Code of switching"
BASE_CODE_NIGHT="Code for switching Day"
BASE_CODE_DAY="Code for switching Night"

BASE_KEYB_ONPOSITION="Keyboard to position"

BASE_KEYB_MODE="Mode of keyboard"
BASE_KEYB_MODE_DIRECT="Direct choice number (phone)"
BASE_KEYB_MODE_MEMORY="Choice of number from memory"
BASE_BACKLIGHT_MODE="Backlight mode"
BASE_BACKLIGHT_MODE_OFF="Off"
BASE_BACKLIGHT_MODE_AUTO="Auto"
BASE_SENSOR_TIT="Door sensors"
BASE_SENSOR1_TIT="Sensor 1"
BASE_SENSOR2_TIT="Sensor 2"

GUARD_LOAD_FAIL="Guard setting load failed"
GUARD_SAVE_FAIL="Guard setting save failed"
GUARDSTAT_ERR="Guard status loading error. Status has not been loaded completely."

#pametova tlacitka
MEM_BUTT="Button"
MEM_NIGHT="Group NIGHT"
MEM_DAY="Group DAY"

#gsm gate
GSMBR_TIT="GSM Gate"
GSMBR_DISA_TIT="DISA"
GSMBR_OGM_TIT="OGM"
GSMBR_REC_TIT="rec"
GSMBR_OPWAIT_TIT="Wait"
GSMBR_ERSCLP_TIT="Erase CLIP"
GSMBR_AEC_TIT="Echo canceller"
GSMBR_LOAD_FAIL="GSM gate parameters loading error"
GSMBR_SAVE_FAIL="GSM gate parameters save error"
GSMBR_0BCLP_TIT="Add zero before CLIP"

#gsm brana - kanal
GSMX_TIT="GSM Gate - module"
GSMX_PIN_TIT="PIN"
GSMX_MSN_TIT="Subscriber number"
GSMX_CALLCHARG_TIT="Call charge"
GSMX_PULSE_EACH_TIT="Pulse each"
GSMX_VOL_GSM_TIT="Volume GSM"
GSMX_VOL_ISDN_TIT="Volume VoIP"
GSMX_CALLIN_TIT="Incoming calls"
GSMX_CALLOUT_TIT="Outgoing calls"
GSMX_PROGRESS_TIT="Call progress tones"
GSMX_SMCALLBACK_TIT="Smart callback"
GSMX_0_TIT="0"
GSMX_CLIR_TIT="CLIR"
GSMX_ROAMING_TIT="Roaming"

#gsm brana - povolene smery
GSMXDIR_TIT="Permitted calls - module"

#gsm brana - volba modulu
MENU_GSMMOD="Auto routing"
GSMMOD_TIT="Automatic routing"
GSMMOD_ENABLE_TIT="Enable automatic routing"
GSMMOD_CALLED_TIT="Called number"
GSMMOD_START_TIT="min"
GSMMOD_END_TIT="max"
GSMMOD_FIRST_TIT="Use module"
GSMMOD_SWAP1_TIT="Swaping"
GSMMOD_SWAP2_TIT="modules"
GSMMOD_FALLOVER1_TIT="Enable"
GSMMOD_FALLOVER2_TIT="fallover"

#gsm brana - prirazeni cisel
GSMNUM_TIT="Subscriber numbers table"
GSMNUM_NUM="Number"
GSMNUM_ADDR="IP address"

#gsm brana - status
MENU_GSMSTAT="GSM Gate status"
GSMSTAT_TIT="GSM Gate status"
GSMSTAT_MODULE_TIT="Module"
GSMSTAT_CHANNEL_TIT="Frequency channel"
GSMSTAT_STRENGTH_TIT="Signal strength"
GSMSTAT_BCCH_TIT="BCCH strength"
GSMSTAT_COUNTRY_TIT="Country code"
GSMSTAT_NETCODE_TIT="Network code"
GSMSTAT_AREA_TIT="Area code"
GSMSTAT_CELLID_TIT="Cell ID"
GSMSTAT_IMSI_TIT="IMSI"
GSMSTAT_ERRRATE_TIT="Bit error rate"
GSMSTAT_IMEI_TIT="IMEI"
GSMSTAT_OPERATOR_TIT="Operator"
GSMSTAT_SWVERSION_TIT="GSM Firmware version"
GSMSTAT_ERR="Parameters load error, not all paramaters has been load successfully."

#sip parametry
SIP_TIT="SIP parameters"
SIP_SERVER_TIT="SIP proxy server"
SIP_SERVER_IP_TIT="Address"
SIP_SERVER_PORT_TIT="Port"
SIP_SERVICE_TIT="Server"
SIP_REGISTRAR_TIT="SIP registrar server"
SIP_ROUTE_TIT="Outbound proxy"
SIP_ACCOUNT_TIT="Account"
SIP_ACCOUNT_MOD_TIT="Account module "
SIP_ACCOUNT_NAME_TIT="Name"
SIP_ACCOUNT_PASS_TIT="Password"
SIP_ACCOUNT_AUTHID_TIT="Auth. Id"
SIP_ACCOUNT_EXPIRE_TIT="Expiration [sec]"
SIP_REG_PROGRESS="Registration in progress"
SIP_REG_SUCCESS="Registration successful"
SIP_REG_FAIL="Registration failed"
SIP_PROVIS_180_TIT="Use (180 Ringing)"
SIP_PROVIS_183_TIT="Use (183 Session progress)"
SIP_SYMMETRIC_RTP_TIT="Enable Simmetric RTP"
SIP_DTMF_EVENT_SIP_TIT="Send DTMF as SIP INFO"
SIP_DTMF_EVENT_RTP_TIT="Send DTMF according to RFC2833"

#services
ADMIN_TIT="Admin services"
UPLOAD_VERSION="VoIP version"
UPLOAD_GUARD_VERSION="UDV version"
DOWNLOAD_LOGFILE="Download log file"
ADMIN_SHOW_LOGCALL="Show call log"
ADMIN_SHOW_LOGREG="Show register log"
ADMIN_SHOW_LOGVOIP="Show VoIP log"
ADMIN_ENABLE_LOG="start enhanced log"
ADMIN_DISABLE_LOG="stop enhanced log"
ADMIN_NTP_SERVER="Time server"
ADMIN_SYSLOG_SERVER="Syslog server"
ADMIN_FW_FILE="Firmware upgrade"
ADMIN_FW_PATIENCE="Upgrade take about 4 minutes. The 'restart' button will show up after finish."
ADMIN_LANG_FILE="Upload language"
ADMIN_CONF_SAVE="Save configuration"
ADMIN_CONF_LOAD="Upload configuration"
ADMIN_FW_SUBMIT="save"
ADMIN_RESTART="restart"
ADMIN_WAIT_RESTART_TIT="Waiting for restart ..."
ADMIN_PASSWD_TIT="Service password"
ADMIN_PASSWD2_TIT="Retype password"
ADMIN_PASSWD_DIFFER="Invalid password confirmation!"

#sip server
SIPSRV_TIT="SIP Server"
SIPSRV_CLR_SIP_SETTING="Before enable internal SIP server you must disable use of external SIP server in SIP parameters menu."
SIPSRV_ENABLE_TIT="Enable SIP server"
SIPSRV_REALM_TIT="Server name (realm)"
SIPSRV_PREFIX_TIT="Prefix"
SIPSRV_USER_TIT="Number"
SIPSRV_PASS_TIT="Password"
SIPSRV_USER_CONN="Connected"

#den noc
MENU_DAYNIGHT="Day Intervals"
DAYNIGHT_INTERVAL_TIT="Interval "
DAYNIGHT_COMMENT="Empty day line is considered as full day.<br>All outside entered intervals in one day is considered as night."
WEEK_START="SUN"
DOW_MON="Mon"
DOW_TUE="Tue"
DOW_WED="Wed"
DOW_THU="Thu"
DOW_FRI="Fri"
DOW_SAT="Sat"
DOW_SUN="Sun"

#curr time
CURRTIME_NOTSET="Time is not set"
CURRTIME_NTP_NOTSET="Time server is not set"

#config load
LOAD_CONF_TIT="Configuration loading..."
LOAD_CONF_SUCCESS="Configuration seccessfully loaded"
LOAD_CONF_ERROR="Configuration loading error"

#user interface
MENU_UI="User interface"
UI_VID_WWW_ENABLE="Video on start page"
UI_VID_PASSWD_ENABLE="Protect video by password"
UI_VID_PASSWD_ALERT="Only web browser will be able to display video"
UI_VID_SURVEILLANCE_ENABLE="Video surveillance (H.264)"
UI_VID_SNOM_PUSH="Push video"
UI_HTTP_PORT="Web interface tcp port"
UI_TELNET_ENABLE="Enable telnet"

#snmp
MENU_SNMP="Setting SNMP"
SENSOR_OPEN_TIMEOUT_TIT="Sensor open timeout"
SNMP_ENABLE_TIT="Enable SNMP"
SNMP_COMM_TIT="Community"
SNMP_COMM_ADDR_TIT="Admin address"
SNMP_VARS_TIT="Variables in MIB tree"
MIB_STARTUP_TIT="Startup"
MIB_SENSOR_OPEN_TIT="Open sensor"
MIB_RELAY_OPEN_TIT="Open relay"
MIB_SENSOR_TIMEOUT_TIT="Open sensor too long"
MIB_KEY_MISMATCH_TIT="Entered invalid keycode"


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


HELP_NET_GUARD="
<ul>
<li><b>Hostname</b> is name of doorphone for resolution in nets (e.g . while using more doorphones, more entrance)
    More using for line identification in SIP - Hostname = number of self telephone number.
    Will appear in title of web browser if connected to guard.
<li><b>Setup via DHCP</b> is used for automatic host configuration.
<li><b>IP address</b> is address used for connection to guard
<li><b>Network gateway and DNS</b> servers are needed only if you make connections or register via internet.
    If guard is used only on local network, leave empty.
<li>Network gateway usually <u>is not address</u> of SIP server (PBX).
</ul>
"

HELP_NET_GSM="
<ul>
<li><b>Select mode</b>
    <ul>
    <li>Two-channels GSM gateway - calls are automatically routed via first or second GSM channel.
        Which channel will be chosen depends on entries in menu <a href="#HELP_GSMMOD">Auto routing</a>
    <li>Two independent GSM gateways - calls incomming to gateway first IP address are routed to first GSM module
        and calls from GSM to first GSM channel (first SIM card number) output from gateway with first IP address
        as source address. Same is valid for second IP address and second GSM channel.
        So there are two complete independent VoIP-GSM gateways each with its own IP address and GSM number
    <li>Single-channel GSM gateway - disable second GSM channel and second VoIP address,
        so it switches to simple one-channel VoIP-GSM gateway
    </ul>
<li><b>Setup via DHCP</b> is used for automatic host configuration. There is need properly setup IDs and DHCP server
    to be able assign one to three IP addres (depends on gateway setup) to one ethernet interface.
<li><b>IP address 1 and 2</b> are addresses of GSM gates.<br>
    In P2P mode make calls to these addresses<br>
    If use internal or external SIP server, make calls to address of SIP server
<li><b>IP address SIP server</b> is used for internal SIP server,
    if enabled all clients (phones,extensions) must register itself and make calls via this address
<li><b>Network gateway and DNS</b> servers are needed only if you make connections or register via internet.
    If GSM Gateway is used only on local network, leave empty.
<li>Network gateway usually <u>is not address</u> of internal or external SIP server.
</ul>
"

HELP_AUDIO="
<ul>
<li>in case of poor audio quality try another codec priority
<li>in menu 'Setting audio' choose G711&micro; as first, G711a as second, ...
<li>check client (IP Phone ) audio settings. At least G711&micro; or G711a must be enabled
</ul>

"
HELP_VIDEO="
<ul>
<li>if you have slow network with long latence, try set frames per second to lower value
    and/or smaller image size
</ul>
"

HELP_GSMBR="
<ul>
<li><b><a name="HELP_GSMBR_DISA"></a>DISA</b>,
    if a number of digits in the direct dialling code are being set, the number will be higher
    than zero, the callers may dial directly using tone selection to the required extension.    
<li><b>OGM</b>
    The gate can be provided additionally with a board with a digital
    recorder of 'messages' for individual GSM channels. When collecting an
    incoming call through direct dialing, these messages are always replayed.
<li><b>Wait</b> sets, how many seconds the automatic operator waits for a
    pre-selected number for direct dialling code selection.
<li><b>Erase Clip</b> is parameter for deleting the initial digits from CLIP. 
</ul>
"

HELP_GSMX="
<ul>
<li><b>PIN</b> for SIM card inserted in GSM module.
<li><b>Subscriber number</b> is used as target of all calls from GSM not resolver by
    <a href="#HELP_GSMBR_DISA">DISA</a> or <a href="#HELP_GSMX_SMCALLBACK">smart callback</a>.
    User should insert this number to subscribed users in internal <a href="#HELP_SIPSRV">Sip server</a> if enabled
    or into your external Sip server (PBX).
    In case of P2P mode (don't use any Sip server) you should enter Subscriber number in
    <a href="#HELP_GSMNUM">Number translation</a> table.
    If this is not setup properly, GSM gateway do not know where to route GSM calls so
    these calls <b>will not function</b>!
<li><b>Volume GSM, Volume VoIP</b> allows adjusting call volume in both directions.
<li><b>Incoming calls, Outgoing calls</b> is permission of incoming/outgoing calls.
<li><b>Call progress tones</b> is signalling of searching of called part in the GSM network.
<li><b><a name="HELP_GSMX_SMCALLBACK"></a>Smart callback</b> 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><b>0, CLIR</b>. It adds 0 or code for Switch OFF outgoing CLIP before each outgoing dialled number.
<li><b>Roaming</b> enable/disable roaming.
</ul>
"

HELP_GSMMOD="
<ul>
<li>Can be set to do LCR (low cost routing)
<li>Fill the table by prefixes of called numbers. In right columns check which GSM module use first
for routing your call and if fallback to second module is possible in case of first module is busy or swapping of modules
is required.
Line with the longest match with called number is selected.
If table is left empty, all directions are permitted and gsm modules are for outgoing calls swapped regularly.
</ul>
"

HELP_GSMNUM="
<ul>
<li>Associate IP adresses with extension numbers. At least is need to enter
    <a href="#HELP_GSMX">Subscriber numbers</a> of GSM channels on which the GSM gate will route incomming calls.
<li>Before IP address you can prepend subscriber number followed by '@'. For example '123@192.168.1.30'.
    Some VoIP phones requires subscriber number before IP address, others deny.
</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><b>Mode</b> of DoorPhone choice selects number per Day/Night DoorPhone mode or selects numbers of the first and 
    second groups. 
<li><b>Sign</b> 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><b>Command for DAY / NIGHT</b> mode switching
    <i>Note: The switchover to Day/Night mode remains set in guard even after power supply disconnection.</i>
<li><b>Automatic or manual switching Day - Night</b> mode. Automatic switch setting in 'Day interface'.
<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><b>Relay mode:</b><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><b>Password for relay closing</b> 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><b>Command</b> 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><b>Duration</b> 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><b>Time</b> 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><b>Max. time</b>, for which the DoorPhone is hanging up, this time can be extended during call by sign choice from 
    telephone (* or #)
<li><b>Number</b> 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><b>Max. time</b> [sec] among button presses [range 1-9]<br>
    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><b>Time</b> [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 <b>Day mode</b>. 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 <b>Night mode</b>. 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><b>SIP proxy server</b> - here enter IP address and port (if differ from default 5060)
    of server to route calls to (PBX, Outbound proxy).
<li><b>SIP registrar server</b> is used for registration only, in most cases is same as SIP proxy server
    and there is not need to enter it. If you don't enter any, system will register to SIP proxy server.
<li><b>Proxy server</b> enter only if you must go to SIP server and Registration server through
    another (proxy) server. Usually not needed, so leave empty.
<li><b>Name and password</b> are not mandatory but must be set axactly same as on your SIP server machine
<li>after entered data saved, registration attempt is executed (if name is not empty) and result shown
<li>if the registration fails, you will see reason in Registration log in menu <a href="#HELP_ADMIN">Service</a>.
<li><b>Choose 180 or 183</b> preanswer signalling. Usually works and not need to change it.
<li><b>Symmetric RTP</b> enable in case of voice flow only in one direction. Usually needed by Cisco devices.
</ul>
"

HELP_SIPSRV="
<ul>
<li>The <b>Realm</b> is name send from SIP server to client as the SIP server name at registration or call build.
<li><b>Prefixes</b> are mandatory only in two channel gateway in 'Two independent GSM gateways' mode.
    In all other modes is use of prefixes your choice.
    Prefix 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 if used must not be the same.
    If leaved empty, internal SIP server will all non-local calls (not for local subscribers) route to GSM.
<li>If automatic routing is enabled, only prefix 1 is in use
<li><b>Client numbers and passwords</b> must be entered same as in extensions settings,
    in other case extesions will not be able to register nor make a call
<li>After successfull extension registration the 'Connected' will appear in right column.
</ul>
"

HELP_ADMIN="
<ul>
<li><b>Download log file</b>. If you are in troubles and need technical support, you will need this file. Follow these steps:
  <ol>
  <li>press 'start 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><b>Show call log</b><br>
    - record of few last incomming and outgoing calls. In case of error, reason is also shown.
<li><b>Show registration log</b><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>Click on <b>'Show VoIP log'</b> start VoIP monitor - log file which is running in new browser window - online displays events.
<li><b>Time server</b> is adress of NTP server with time data (actual time in module VoIP is displayed in 'Day intervals') 
      if doesn't know address of NTP, you use asterisk '*' and system choose acceptable one.
<li><b>Syslog server</b> is computer able of receive internal messages from device
<li>In <b>Firmware upgrade</b> 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 <b>unusable!</b><br>
    After firmware upgrade pressing reset button is allways required.
<li><b>Upload language</b> - possibility to supply custom language.<br>
  Name of this file appears in Language combo box. HTML entities in file name are allowed.<br>
  Language file must start with line cs or en - the language from which thist customization is derived.
<li><b>Save configuration</b> - save actual setting in IPDP (all features)
<li><b>Restore configuration</b> - restores setting of all IPDP from file previously storage configuration
<li><b>Service password</b>, change the manufacturer default password is highly recommended.
</ul>
"

HELP_DAYNIGHT="
<ul>
<li>Display only if you check automatic switch Day/Night mode on <a href="#HELP_VRATNY">Basic parameter</a>
<li>Display actual internal time in right up corner (you must set Time server (NTP) in <a href="#HELP_ADMIN">'Service'</a> menu)
<li>Table of time interval - interval is meaning where is day, the rest is night. For example: 
    interval 1 = 08:00-12:00, interval 
             2 = 14:00-17:00 
        then from 00:00 to 7:59 is night, from 8:00 to 12:00 is day, from 12:01 to 13:59 is night, 
        from 14:00 to 17:00 is day and from 17:01 to 23:59 is night.
</ul>
"

HELP_UI_GSM="
<ul>
<li>Possibility change port from conventional 80 to other
<li>Possibility disable / enable access over telnet 
</ul>
"

HELP_UI_GUARD="
<ul>
<li>Possibility switch off displaying videos on start page - safety device
<li>Possibility switch off video in VoIP call (video in call makes at some systems problem) 
<li>Possibility switch off video if calling is not in progress (surveillance)
<li>Possibility switch on function 'Push video' in Snom phones
<li>Possibility change port from conventional 80 to other
<li>Possibility disable / enable access over telnet 
</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
<a name="HELP_LOGFILE"></a>log file downloadable from menu <a href="#HELP_ADMIN">Admin</a>.
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 <a href="#HELP_ADMIN">Service</a> 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 in menu <a href="#HELP_SIP">'SIP parameters'</a>.<br>
    - check network connection and if your SIP server (PBX) is running

<li>registration is unsuccessful - in <a href="#HELP_ADMIN">registration log</a> is reason: 404 (Not found)<br>
    - check IP address of SIP server, port and registration name in menu <a href="#HELP_SIP">'SIP parameters'</a>

<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 <a href="#HELP_ADMIN">Service</a> download log file and send it
      with problem description to technical support
</ul>

<h3>Call</h3>
<ul>
<li>build the call is not possible<br>
    - in menu <a href="#HELP_ADMIN">Service</a> 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 improperly 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 <a href="#HELP_SIP">'SIP Setting'</a>
<li>in call log I see 'Unsupported media type'<br>
    - in <a href="#HELP_AUDIO">'Setting audio'</a> 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 <a href="#HELP_ADMIN">Service</a> download log file and send it with description to technical support
</ul>

<h3>Audio</h3>
<ul>
<li>poor quality<br>
    - in <a href="#HELP_AUDIO">'Setting audio'</a> choose as priority one codec G711&micro;, second G711a and so on<br>
    - check extension 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 <a href="#HELP_LOGFILE">log file</a> from menu 'Service' send to technical support
"
