Changeset 7856

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Timestamp:
06/21/09 05:17:22 (1 year ago)
Author:
mickecarlsson
Message:

Fixed spelling errors, updated sipsettings.pot

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  • modules/branches/2.6/sipsettings/i18n/sipsettings.pot

    r7815 r7856  
    2222"Project-Id-Version: PACKAGE VERSION\n" 
    2323"Report-Msgid-Bugs-To: \n" 
    24 "POT-Creation-Date: 2009-06-14 11:26+0200\n" 
     24"POT-Creation-Date: 2009-06-21 14:16+0200\n" 
    2525"PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n" 
    2626"Last-Translator: FULL NAME <EMAIL@ADDRESS>\n" 
     
    3030"Content-Transfer-Encoding: 8bit\n" 
    3131 
    32 #: natget.html.php:33 natget.html.php:70 
    33 msgid "Failed to auto-detect settinggs" 
     32#: functions.inc.php:357 
     33#, php-format 
     34msgid "%s must be a non-negative integer" 
     35msgstr "" 
     36 
     37#: functions.inc.php:361 
     38msgid "Bind Address (bindaddr) must be an IP address." 
     39msgstr "" 
     40 
     41#: functions.inc.php:366 
     42msgid "Bind Port (bindport) must be between 1024..65535, default 5060" 
     43msgstr "" 
     44 
     45#: functions.inc.php:374 
     46msgid "rtpholdtimeout must be higher than rtptimeout" 
     47msgstr "" 
     48 
     49#: functions.inc.php:405 
     50#, php-format 
     51msgid "%s must be alphanumeric" 
     52msgstr "" 
     53 
     54#: functions.inc.php:420 
     55msgid "External IP can not be blank" 
     56msgstr "" 
     57 
     58#: functions.inc.php:428 
     59msgid "Dynamic Host can not be blank" 
     60msgstr "" 
     61 
     62#: functions.inc.php:466 
     63msgid "Localnet setting must be an IP address" 
     64msgstr "" 
     65 
     66#: functions.inc.php:471 
     67msgid "Localnet netmask must be formated properly (e.g. 255.255.255.0 or 24)" 
     68msgstr "" 
     69 
     70#: natget.html.php:33 
     71msgid "Failed to auto-detect settings" 
    3472msgstr "" 
    3573 
     
    3876msgstr "" 
    3977 
    40 #: page.sipsettings.php:39 
     78#: natget.html.php:68 
     79msgid "Failed to auto-detect settinggs" 
     80msgstr "" 
     81 
     82#: page.sipsettings.php:141 
    4183msgid "Edit Settings" 
    4284msgstr "" 
    4385 
    44 #: page.sipsettings.php:132 
    45 msgid "General Settings" 
    46 msgstr "" 
    47  
    48 #: page.sipsettings.php:136 
    49 msgid "Language" 
    50 msgstr "" 
    51  
    52 #: page.sipsettings.php:136 
    53 msgid "Default Language for a channel, Asterisk: language" 
    54 msgstr "" 
    55  
    56 #: page.sipsettings.php:143 
    57 msgid "SRV Lookup" 
    58 msgstr "" 
    59  
    60 #: page.sipsettings.php:143 
    61 msgid "" 
    62 "Enable Asterisk srvlookup. See current version of Asterisk for limitations " 
    63 "on SRV functionality." 
    64 msgstr "" 
    65  
    66 #: page.sipsettings.php:150 
    67 msgid "Enabled" 
    68 msgstr "" 
    69  
    70 #: page.sipsettings.php:155 
    71 msgid "Disabled" 
    72 msgstr "" 
    73  
    74 #: page.sipsettings.php:165 
    75 msgid "Notification & MWI" 
    76 msgstr "" 
    77  
    78 #: page.sipsettings.php:170 
    79 msgid "MWI Polling Freq" 
    80 msgstr "" 
    81  
    82 #: page.sipsettings.php:170 
    83 msgid "" 
    84 "Frequency in seconds to check if MWI state has changed and inform peers." 
    85 msgstr "" 
    86  
    87 #: page.sipsettings.php:177 
    88 msgid "Notify Ringing" 
    89 msgstr "" 
    90  
    91 #: page.sipsettings.php:177 
    92 msgid "" 
    93 "Control whether subscriptions already INUSE get sent RINGING when another " 
    94 "call is sent. Useful when using BLF." 
    95 msgstr "" 
    96  
    97 #: page.sipsettings.php:184 page.sipsettings.php:207 page.sipsettings.php:262 
    98 #: page.sipsettings.php:285 page.sipsettings.php:454 
    99 msgid "Yes" 
    100 msgstr "" 
    101  
    102 #: page.sipsettings.php:189 page.sipsettings.php:212 page.sipsettings.php:267 
    103 #: page.sipsettings.php:290 page.sipsettings.php:459 
    104 msgid "No" 
    105 msgstr "" 
    106  
    107 #: page.sipsettings.php:200 
    108 msgid "Notify Hold" 
    109 msgstr "" 
    110  
    111 #: page.sipsettings.php:200 
    112 msgid "" 
    113 "Control whether subscriptions INUSE get sent ONHOLD when call is placed on " 
    114 "hold. Useful when using BLF." 
    115 msgstr "" 
    116  
    117 #: page.sipsettings.php:222 
     86#: page.sipsettings.php:169 
     87msgid "ERRORS" 
     88msgstr "" 
     89 
     90#: page.sipsettings.php:185 
     91msgid "NAT Settings" 
     92msgstr "" 
     93 
     94#: page.sipsettings.php:190 
     95msgid "Nat" 
     96msgstr "" 
     97 
     98#: page.sipsettings.php:190 
     99msgid "" 
     100"Asterisk nat setting:<br /> yes = Always ignore info and assume NAT<br /> no " 
     101"= Use NAT mode only according to RFC3581 <br /> never = Never attempt NAT " 
     102"mode or RFC3581 <br /> route = Assume NAT, don't send rport" 
     103msgstr "" 
     104 
     105#: page.sipsettings.php:218 
     106msgid "IP Configuration" 
     107msgstr "" 
     108 
     109#: page.sipsettings.php:218 
     110msgid "" 
     111"Indicate whether the box has a public IP or requires NAT settings. Automatic " 
     112"configuration of what is often put in sip_nat.conf" 
     113msgstr "" 
     114 
     115#: page.sipsettings.php:225 
     116msgid "Public IP" 
     117msgstr "" 
     118 
     119#: page.sipsettings.php:228 
     120msgid "Static IP" 
     121msgstr "" 
     122 
     123#: page.sipsettings.php:231 
     124msgid "Dynamic IP" 
     125msgstr "" 
     126 
     127#: page.sipsettings.php:239 
     128msgid "External IP" 
     129msgstr "" 
     130 
     131#: page.sipsettings.php:239 
     132msgid "" 
     133"External Static IP or FQDN as seen on the WAN side of the router. (asterisk: " 
     134"externip)" 
     135msgstr "" 
     136 
     137#: page.sipsettings.php:245 
     138msgid "Dynamic Host" 
     139msgstr "" 
     140 
     141#: page.sipsettings.php:245 
     142msgid "" 
     143"External FQDN as seen on the WAN side of the router and updated dynamically, " 
     144"e.g. mydomain.dyndns.com. (asterisk: externhost)" 
     145msgstr "" 
     146 
     147#: page.sipsettings.php:250 
     148msgid "Refresh Rate" 
     149msgstr "" 
     150 
     151#: page.sipsettings.php:250 
     152msgid "" 
     153"Asterisk: externrefresh. How often to lookup and refresh the External Host " 
     154"FQDN, in seconds." 
     155msgstr "" 
     156 
     157#: page.sipsettings.php:255 
     158msgid "Local Networks" 
     159msgstr "" 
     160 
     161#: page.sipsettings.php:255 
     162msgid "" 
     163"Local network settings (Asterisk: localnet) in the form of ip/mask such as " 
     164"192.168.1.0/255.255.255.0. For networks with more 1 lan subnets, use the Add " 
     165"Local Network Field button for more fields. Blank fields will be removed " 
     166"upon submitting." 
     167msgstr "" 
     168 
     169#: page.sipsettings.php:299 
    118170msgid "Audio Codecs" 
    119171msgstr "" 
    120172 
    121 #: page.sipsettings.php:225 
     173#: page.sipsettings.php:302 
    122174msgid "Codecs" 
    123175msgstr "" 
    124176 
    125 #: page.sipsettings.php:225 
     177#: page.sipsettings.php:302 
    126178msgid "" 
    127179"Check the desired codecs, all others will be disabled unless explicitly " 
     
    129181msgstr "" 
    130182 
    131 #: page.sipsettings.php:255 
     183#: page.sipsettings.php:332 
    132184msgid "Non-Standard g726" 
    133185msgstr "" 
    134186 
    135 #: page.sipsettings.php:255 
     187#: page.sipsettings.php:332 
    136188msgid "" 
    137189"Asterisk: g726nonstandard. If the peer negotiates G726-32 audio, use AAL2 " 
     
    141193msgstr "" 
    142194 
    143 #: page.sipsettings.php:278 
     195#: page.sipsettings.php:339 page.sipsettings.php:360 page.sipsettings.php:445 
     196#: page.sipsettings.php:495 page.sipsettings.php:516 page.sipsettings.php:585 
     197#: page.sipsettings.php:698 
     198msgid "Yes" 
     199msgstr "" 
     200 
     201#: page.sipsettings.php:343 page.sipsettings.php:364 page.sipsettings.php:449 
     202#: page.sipsettings.php:499 page.sipsettings.php:520 page.sipsettings.php:589 
     203#: page.sipsettings.php:702 
     204msgid "No" 
     205msgstr "" 
     206 
     207#: page.sipsettings.php:353 
    144208msgid "T38 Pass-Through" 
    145209msgstr "" 
    146210 
    147 #: page.sipsettings.php:278 
     211#: page.sipsettings.php:353 
    148212msgid "" 
    149213"Asterisk: t38pt_udptl. Enables T38 passthrough if enabled. This SIP channels " 
     
    152216msgstr "" 
    153217 
    154 #: page.sipsettings.php:300 
     218#: page.sipsettings.php:373 
    155219msgid "Video Codecs" 
    156220msgstr "" 
    157221 
    158 #: page.sipsettings.php:304 
     222#: page.sipsettings.php:378 
    159223msgid "Video Support" 
    160224msgstr "" 
    161225 
    162 #: page.sipsettings.php:304 
     226#: page.sipsettings.php:378 
    163227msgid "Check to enable and then choose allowed codecs." 
    164228msgstr "" 
    165229 
    166 #: page.sipsettings.php:308 
    167 msgid "enable" 
    168 msgstr "" 
    169  
    170 #: page.sipsettings.php:343 
     230#: page.sipsettings.php:385 page.sipsettings.php:564 page.sipsettings.php:719 
     231msgid "Enabled" 
     232msgstr "" 
     233 
     234#: page.sipsettings.php:389 page.sipsettings.php:568 page.sipsettings.php:723 
     235msgid "Disabled" 
     236msgstr "" 
     237 
     238#: page.sipsettings.php:427 
    171239msgid "Max Bit Rate" 
    172240msgstr "" 
    173241 
    174 #: page.sipsettings.php:343 
     242#: page.sipsettings.php:427 
    175243msgid "Maximum bitrate for video calls in kb/s" 
    176244msgstr "" 
    177245 
    178 #: page.sipsettings.php:349 
    179 msgid "NAT Settings" 
    180 msgstr "" 
    181  
    182 #: page.sipsettings.php:354 
    183 msgid "Nat" 
    184 msgstr "" 
    185  
    186 #: page.sipsettings.php:354 
    187 msgid "" 
    188 "Asterisk nat setting:<br /> yes = Always ignore info and assume NAT<br /> no " 
    189 "= Use NAT mode only according to RFC3581 <br /> never = Never attempt NAT " 
    190 "mode or RFC3581 <br /> route = Assume NAT, don't send rport" 
    191 msgstr "" 
    192  
    193 #: page.sipsettings.php:385 
    194 msgid "IP Configuration" 
    195 msgstr "" 
    196  
    197 #: page.sipsettings.php:385 
    198 msgid "" 
    199 "Indicate whether the box has a public IP or requires NAT settings. Automatic " 
    200 "onfiguration of what is often put in sip_nat.conf" 
    201 msgstr "" 
    202  
    203 #: page.sipsettings.php:392 
    204 msgid "Public" 
    205 msgstr "" 
    206  
    207 #: page.sipsettings.php:395 
    208 msgid "Static" 
    209 msgstr "" 
    210  
    211 #: page.sipsettings.php:398 
    212 msgid "Dynamic" 
    213 msgstr "" 
    214  
    215 #: page.sipsettings.php:401 
    216 msgid "Auto Configure" 
    217 msgstr "" 
    218  
    219 #: page.sipsettings.php:409 
    220 msgid "External IP" 
    221 msgstr "" 
    222  
    223 #: page.sipsettings.php:409 
    224 msgid "" 
    225 "External Static IP or FQDN as seen on the WAN side of the router. (asterisk: " 
    226 "externip)" 
    227 msgstr "" 
    228  
    229 #: page.sipsettings.php:415 
    230 msgid "Dynamic Host" 
    231 msgstr "" 
    232  
    233 #: page.sipsettings.php:415 
    234 msgid "" 
    235 "External FQDN as seen on the WAN side of the router and updated dynamically, " 
    236 "e.g. mydomain.dyndns.com. (asterisk: externhost)" 
    237 msgstr "" 
    238  
    239 #: page.sipsettings.php:420 
    240 msgid "External Refresh" 
    241 msgstr "" 
    242  
    243 #: page.sipsettings.php:420 
    244 msgid "How often to refresh the External Host FQDN." 
    245 msgstr "" 
    246  
    247 #: page.sipsettings.php:425 
    248 msgid "Local Neworks" 
    249 msgstr "" 
    250  
    251 #: page.sipsettings.php:425 
    252 msgid "" 
    253 "Local network settings (Asterisk: localnet) in the form of ip/mask such as " 
    254 "192.168.1.0/255.255.255.0. For networks with more than 2 lan subnets, use " 
    255 "the Additional SIP settings below to define them." 
    256 msgstr "" 
    257  
    258 #: page.sipsettings.php:442 
     246#: page.sipsettings.php:429 
     247msgid "kb/s" 
     248msgstr "" 
     249 
     250#: page.sipsettings.php:433 
    259251msgid "MEDIA & RTP Settings" 
    260252msgstr "" 
    261253 
    262 #: page.sipsettings.php:447 
     254#: page.sipsettings.php:438 
    263255msgid "Reinvite Behavior" 
    264256msgstr "" 
    265257 
    266 #: page.sipsettings.php:447 
     258#: page.sipsettings.php:438 
    267259msgid "" 
    268260"Asterisk: canreinvite. yes: standard reinvites; no: never; nonat: An " 
     
    274266msgstr "" 
    275267 
    276 #: page.sipsettings.php:478 
     268#: page.sipsettings.php:466 
    277269msgid "RTP Timers" 
    278270msgstr "" 
    279271 
    280 #: page.sipsettings.php:478 
     272#: page.sipsettings.php:466 
    281273msgid "" 
    282274"Asterisk: rtptimeout. Terminate call if rtptimeout seconds of no RTP or RTCP " 
    283275"activity on the audio channel when we're not on hold. This is to be able to " 
    284276"hangup a call in the case of a phone disappearing from the net, like a " 
    285 "powerloss or grandma tripping over a cable.<br /> Asterisk: rtpholdtimeout. " 
     277"powerloss or someone tripping over a cable.<br /> Asterisk: rtpholdtimeout. " 
    286278"Terminate call if rtpholdtimeout seconds of no RTP or RTCP activity on the " 
    287279"audio channel when we're on hold (must be > rtptimeout). <br /> Asterisk: " 
     
    290282msgstr "" 
    291283 
     284#: page.sipsettings.php:476 
     285msgid "Notification & MWI" 
     286msgstr "" 
     287 
     288#: page.sipsettings.php:481 
     289msgid "MWI Polling Freq" 
     290msgstr "" 
     291 
     292#: page.sipsettings.php:481 
     293msgid "" 
     294"Frequency in seconds to check if MWI state has changed and inform peers." 
     295msgstr "" 
     296 
    292297#: page.sipsettings.php:488 
     298msgid "Notify Ringing" 
     299msgstr "" 
     300 
     301#: page.sipsettings.php:488 
     302msgid "" 
     303"Control whether subscriptions already INUSE get sent RINGING when another " 
     304"call is sent. Useful when using BLF." 
     305msgstr "" 
     306 
     307#: page.sipsettings.php:509 
     308msgid "Notify Hold" 
     309msgstr "" 
     310 
     311#: page.sipsettings.php:509 
     312msgid "" 
     313"Control whether subscriptions INUSE get sent ONHOLD when call is placed on " 
     314"hold. Useful when using BLF." 
     315msgstr "" 
     316 
     317#: page.sipsettings.php:529 
    293318msgid "Registration Settings" 
    294319msgstr "" 
    295320 
    296 #: page.sipsettings.php:493 
    297 msgid "Registration Attempts" 
    298 msgstr "" 
    299  
    300 #: page.sipsettings.php:493 
     321#: page.sipsettings.php:534 
     322msgid "Registrations" 
     323msgstr "" 
     324 
     325#: page.sipsettings.php:534 
    301326msgid "" 
    302327"Asterisk: registertimeout. Retry registration attempts every registertimeout " 
     
    305330"before giving up. A value of 0 means keep trying forever. Normally this " 
    306331"should be set to 0 so that Asterisk will continue to register until " 
    307 "successful in the case of network or gateway outagages." 
    308 msgstr "" 
    309  
    310 #: page.sipsettings.php:503 
     332"successful in the case of network or gateway outages." 
     333msgstr "" 
     334 
     335#: page.sipsettings.php:544 
    311336msgid "Registration Times" 
    312337msgstr "" 
    313338 
    314 #: page.sipsettings.php:503 
     339#: page.sipsettings.php:544 
    315340msgid "" 
    316341"Asterisk: minexpiry. Minimum length of registrations/subscriptions.<br /> " 
     
    320345msgstr "" 
    321346 
    322 #: page.sipsettings.php:540 
     347#: page.sipsettings.php:554 
    323348msgid "Jitter Buffer Settings" 
    324349msgstr "" 
    325350 
    326 #: page.sipsettings.php:561 
     351#: page.sipsettings.php:558 
     352msgid "Jitter Buffer" 
     353msgstr "" 
     354 
     355#: page.sipsettings.php:558 
     356msgid "" 
     357"Asterisk: jbenable. Enables the use of a jitterbuffer on the receiving side " 
     358"of a SIP channel. An enabled jitterbuffer will be used only if the sending " 
     359"side can create and the receiving side can not accept jitter. The SIP " 
     360"channel can accept jitter, thus a jitterbuffer on the receive SIP side will " 
     361"be used only if it is forced and enabled. An example is if receiving from a " 
     362"jittery channel to voicemail, the jitter buffer will be used if enabled. " 
     363"However, it will not be used when sending to a SIP endpoint since they " 
     364"usually have their own jitter buffers. See jbforce to force it's use always." 
     365msgstr "" 
     366 
     367#: page.sipsettings.php:578 
     368msgid "Force Jitter Buffer" 
     369msgstr "" 
     370 
     371#: page.sipsettings.php:578 
     372msgid "" 
     373"Asterisk: jbforce. Forces the use of a jitterbuffer on the receive side of a " 
     374"SIP channel. Normally the jitter buffer will not be used if receiving a " 
     375"jittery channel but sending it off to another channel such as another SIP " 
     376"channel to an endpoint, since there is typically a jitter buffer at the far " 
     377"end. This will force the use of the jitter buffer before sending the stream " 
     378"on. This is not typically desired as it adds additional latency into the " 
     379"stream." 
     380msgstr "" 
     381 
     382#: page.sipsettings.php:599 
     383msgid "Implementation" 
     384msgstr "" 
     385 
     386#: page.sipsettings.php:599 
     387msgid "" 
     388"Asterisk: jbimpl. Jitterbuffer implementation, used on the receiving side of " 
     389"a SIP channel. Two implementations are currently available:<br /> fixed: " 
     390"size always equals to jbmaxsize;<br />) adaptive: with variable size (the " 
     391"new jb of IAX2)." 
     392msgstr "" 
     393 
     394#: page.sipsettings.php:606 
     395msgid "Fixed" 
     396msgstr "" 
     397 
     398#: page.sipsettings.php:610 
     399msgid "Adaptive" 
     400msgstr "" 
     401 
     402#: page.sipsettings.php:620 
     403msgid "Jitter Buffer Logging" 
     404msgstr "" 
     405 
     406#: page.sipsettings.php:620 
     407msgid "Asterisk: jblog. Enables jitter buffer frame logging." 
     408msgstr "" 
     409 
     410#: page.sipsettings.php:627 
     411msgid "Enable" 
     412msgstr "" 
     413 
     414#: page.sipsettings.php:631 
     415msgid "Disable" 
     416msgstr "" 
     417 
     418#: page.sipsettings.php:641 
     419msgid "Jitter Buffer Size" 
     420msgstr "" 
     421 
     422#: page.sipsettings.php:641 
     423msgid "" 
     424"Asterisk: jbmaxsize. Max length of the jitterbuffer in milliseconds.<br /> " 
     425"Asterisk: jbresyncthreshold. Jump in the frame timestamps over which the " 
     426"jitterbuffer is resynchronized. Useful to improve the quality of the voice, " 
     427"with big jumps in/broken timestamps, usually sent from exotic devices and " 
     428"programs. Can be set to -1 to disable." 
     429msgstr "" 
     430 
     431#: page.sipsettings.php:650 
    327432msgid "Advanced General Settings" 
    328433msgstr "" 
    329434 
    330 #: page.sipsettings.php:565 
    331 msgid "Submit Changes" 
    332 msgstr "" 
    333  
    334 #: page.sipsettings.php:597 
    335 msgid "An Error occured trying to fetch Bandwidth.com trunk information" 
    336 msgstr "" 
     435#: page.sipsettings.php:655 
     436msgid "Language" 
     437msgstr "" 
     438 
     439#: page.sipsettings.php:655 
     440msgid "Default Language for a channel, Asterisk: language" 
     441msgstr "" 
     442 
     443#: page.sipsettings.php:664 
     444msgid "Default Context" 
     445msgstr "" 
     446 
     447#: page.sipsettings.php:664 
     448msgid "" 
     449"Asterisk: context. Default context for incoming calls if not specified. " 
     450"FreePBX sets this to from-sip-external which is used in conjunction with the " 
     451"Allow Anonymous SIP calls. If you change this you will effect that behavior. " 
     452"It is recommended to leave this blank." 
     453msgstr "" 
     454 
     455#: page.sipsettings.php:673 
     456msgid "Bind Address" 
     457msgstr "" 
     458 
     459#: page.sipsettings.php:673 
     460msgid "" 
     461"Asterisk: bindaddr. The IP address to bind to and listen for calls on the " 
     462"Bind Port. If set to 0.0.0.0 Asterisk will listen on all addresses. It is " 
     463"recommended to leave this blank." 
     464msgstr "" 
     465 
     466#: page.sipsettings.php:682 
     467msgid "Bind Port" 
     468msgstr "" 
     469 
     470#: page.sipsettings.php:682 
     471msgid "" 
     472"Asterisk: bindport. Local incoming UDP Port that Asterisk will bind to and " 
     473"listen for SIP messages. The SIP standard is 5060 and in most cases this is " 
     474"what you want. It is recommended to leave this blank." 
     475msgstr "" 
     476 
     477#: page.sipsettings.php:691 
     478msgid "Allow SIP Guests" 
     479msgstr "" 
     480 
     481#: page.sipsettings.php:691 
     482msgid "" 
     483"Asterisk: allowguest. When set Asterisk will allow Guest SIP calls and send " 
     484"them to the Default SIP context. Turning this off will keep anonymous SIP " 
     485"calls from entering the system. However, the Allow Anonymous SIP calls from " 
     486"the General Settings section will not function. Allowing guest calls but " 
     487"rejecting the Anonymous SIP calls in the General Section will enable you to " 
     488"see the call attempts and debug incoming calls that may be mis-configured " 
     489"and appearing as guests." 
     490msgstr "" 
     491 
     492#: page.sipsettings.php:712 
     493msgid "SRV Lookup" 
     494msgstr "" 
     495 
     496#: page.sipsettings.php:712 
     497msgid "" 
     498"Enable Asterisk srvlookup. See current version of Asterisk for limitations " 
     499"on SRV functionality." 
     500msgstr "" 
     501 
     502#: page.sipsettings.php:735 
     503msgid "Other SIP Settings" 
     504msgstr "" 
     505 
     506#: page.sipsettings.php:735 
     507msgid "" 
     508"You may set any other SIP settings not present here that are allowed to be " 
     509"configured in the General section of sip.conf. There will be no error " 
     510"checking against these settings so check them carefully. They should be " 
     511"entered as:<br /> [setting] = [value]<br /> in the boxes below. Click the " 
     512"Add Field box to add additional fields. Blank boxes will be deleted when " 
     513"submitted." 
     514msgstr "" 
  • modules/branches/2.6/sipsettings/natget.html.php

    r7822 r7856  
    3131        $fn = "http://mirror.freepbx.org/whatismyip.php"; 
    3232 
    33         $json_array['status'] = _('Failed to auto-detect settinggs'); 
     33        $json_array['status'] = _('Failed to auto-detect settings'); 
    3434        $json_array['externip'] = ''; 
    3535 
  • modules/branches/2.6/sipsettings/page.sipsettings.php

    r7847 r7856  
    216216  <tr> 
    217217    <td> 
    218       <a href="#" class="info"><?php echo _("IP Configuration")?><span><?php echo _("Indicate whether the box has a public IP or requires NAT settings. Automatic onfiguration of what is often put in sip_nat.conf")?></span></a> 
     218      <a href="#" class="info"><?php echo _("IP Configuration")?><span><?php echo _("Indicate whether the box has a public IP or requires NAT settings. Automatic configuration of what is often put in sip_nat.conf")?></span></a> 
    219219    </td> 
    220220    <td> 
     
    467467    </td> 
    468468    <td> 
    469       <input type="text" size="2" id="rtptimeout" name="rtptimeout" class="validate-int" value="<?php echo $rtptimeout ?>" tabindex="<?php echo ++$tabindex;?>"><small>(rtptimeout)</small>&nbsp; 
    470       <input type="text" size="2" id="rtpholdtimeout" name="rtpholdtimeout" class="validate-int" value="<?php echo $rtpholdtimeout ?>" tabindex="<?php echo ++$tabindex;?>"><small>(rtpholdtimeout)</small>&nbsp; 
    471       <input type="text" size="2" id="rtpkeepalive" name="rtpkeepalive" class="validate-int" value="<?php echo $rtpkeepalive ?>" tabindex="<?php echo ++$tabindex;?>"><small>(rtpkeepalive)</small> 
     469      <input type="text" size="2" id="rtptimeout" name="rtptimeout" class="validate-int" value="<?php echo $rtptimeout ?>" tabindex="<?php echo ++$tabindex;?>"><small>&nbsp;(rtptimeout)</small>&nbsp; 
     470      <input type="text" size="2" id="rtpholdtimeout" name="rtpholdtimeout" class="validate-int" value="<?php echo $rtpholdtimeout ?>" tabindex="<?php echo ++$tabindex;?>"><small>&nbsp;(rtpholdtimeout)</small>&nbsp; 
     471      <input type="text" size="2" id="rtpkeepalive" name="rtpkeepalive" class="validate-int" value="<?php echo $rtpkeepalive ?>" tabindex="<?php echo ++$tabindex;?>"><small>&nbsp;(rtpkeepalive)</small> 
    472472    </td> 
    473473  </tr> 
     
    532532  <tr> 
    533533    <td> 
    534       <a href="#" class="info"><?php echo _("Registrations")?><span><?php echo _("Asterisk: registertimeout. Retry registration attempts every registertimeout seconds until successful or until registrationattempts tries have been made.<br /> Asterisk: registrationattempts. Number of times to try and register before giving up. A value of 0 means keep trying forever. Normally this should be set to 0 so that Asterisk will continue to register until successful in the case of network or gateway outagages.")?></span></a> 
    535     </td> 
    536     <td> 
    537       <input type="text" size="2" id="registertimeout" name="registertimeout" class="validate-int" value="<?php echo $registertimeout ?>" tabindex="<?php echo ++$tabindex;?>"><small>(registertimeout)</small>&nbsp; 
    538       <input type="text" size="2" id="registerattempts" name="registerattempts" class="validate-int" value="<?php echo $registerattempts ?>" tabindex="<?php echo ++$tabindex;?>"><small>(registerattempts)</small> 
     534      <a href="#" class="info"><?php echo _("Registrations")?><span><?php echo _("Asterisk: registertimeout. Retry registration attempts every registertimeout seconds until successful or until registrationattempts tries have been made.<br /> Asterisk: registrationattempts. Number of times to try and register before giving up. A value of 0 means keep trying forever. Normally this should be set to 0 so that Asterisk will continue to register until successful in the case of network or gateway outages.")?></span></a> 
     535    </td> 
     536    <td> 
     537      <input type="text" size="2" id="registertimeout" name="registertimeout" class="validate-int" value="<?php echo $registertimeout ?>" tabindex="<?php echo ++$tabindex;?>"><small>&nbsp;(registertimeout)</small>&nbsp; 
     538      <input type="text" size="2" id="registerattempts" name="registerattempts" class="validate-int" value="<?php echo $registerattempts ?>" tabindex="<?php echo ++$tabindex;?>"><small>&nbsp;(registerattempts)</small> 
    539539    </td> 
    540540  </tr> 
     
    671671  <tr> 
    672672    <td> 
    673       <a href="#" class="info"><?php echo _("Bind Address")?><span><?php echo _("Asterisk: bindaddr. The IP adderss to bind to and listen for calls on the Bind Port. If set to 0.0.0.0 Asterisk will listen on all addresses. It is recommended to leave this blank.")?></span></a> 
     673      <a href="#" class="info"><?php echo _("Bind Address")?><span><?php echo _("Asterisk: bindaddr. The IP address to bind to and listen for calls on the Bind Port. If set to 0.0.0.0 Asterisk will listen on all addresses. It is recommended to leave this blank.")?></span></a> 
    674674    </td> 
    675675    <td>