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Diffstat (limited to 'po/sk/pbx.po')
| -rw-r--r-- | po/sk/pbx.po | 484 |
1 files changed, 0 insertions, 484 deletions
diff --git a/po/sk/pbx.po b/po/sk/pbx.po deleted file mode 100644 index 7b6d4a5c6..000000000 --- a/po/sk/pbx.po +++ /dev/null @@ -1,484 +0,0 @@ -msgid "" -msgstr "" -"Content-Type: text/plain; charset=UTF-8\n" -"Project-Id-Version: PACKAGE VERSION\n" -"Last-Translator: Automatically generated\n" -"Language-Team: none\n" -"MIME-Version: 1.0\n" -"Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=(n==1) ? 0 : (n>=2 && n<=4) ? 1 : 2;\n" - -msgid "Advanced Settings" -msgstr "" - -msgid "Available" -msgstr "" - -msgid "" -"Avoid using anything but alpha-numeric characters, space, comma, and period." -msgstr "" - -msgid "Away" -msgstr "" - -msgid "Blacklisted Numbers" -msgstr "" - -msgid "Call Routing" -msgstr "" - -msgid "Call-back Numbers" -msgstr "" - -msgid "Call-back Provider" -msgstr "" - -msgid "Call-through Numbers" -msgstr "" - -msgid "Copy-paste large lists of numbers here." -msgstr "" - -msgid "" -"Designate numbers that are allowed to call through this system and which " -"user's privileges they will have." -msgstr "" - -msgid "" -"Designate numbers to whom the system will hang up and call back, which " -"provider will be used to call them, and which user's privileges will be " -"granted to them." -msgstr "" - -msgid "Dials numbers unmatched elsewhere" -msgstr "" - -msgid "Do Not Disturb" -msgstr "" - -msgid "Domain/IP Address/Dynamic Domain" -msgstr "" - -msgid "Dynamic List of Blacklisted Numbers" -msgstr "" - -msgid "Email" -msgstr "" - -msgid "Enable Incoming Calls (Register via SIP)" -msgstr "" - -msgid "Enable Incoming Calls (set Status below)" -msgstr "" - -msgid "Enable Outgoing Calls" -msgstr "" - -msgid "Enabled" -msgstr "" - -msgid "" -"Enter a VoIP provider to use for call-back in the format username@some.host." -"name, as listed in \"Outgoing Calls\" above. It's easiest to copy and paste " -"the providers from above. Invalid entries, including providers not enabled " -"for outgoing calls, will be rejected silently." -msgstr "" - -msgid "" -"Enter phone numbers that you want to decline calls from automatically. You " -"should probably omit the country code and any leading zeroes, but please " -"experiment to make sure you are blocking numbers from your desired area " -"successfully." -msgstr "" - -msgid "" -"Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices " -"you will use ONLY locally and never from a remote location." -msgstr "" - -msgid "" -"Enter this hostname (or hostname:port) in the Server/Registrar setting of " -"SIP devices you will use from a remote location (they will work locally too)." -msgstr "" - -msgid "External SIP Port" -msgstr "" - -msgid "" -"For each provider enabled for incoming calls, here you can restrict which " -"users to ring on incoming calls. If the list is empty, the system will " -"indicate that all users enabled for incoming calls will ring. Invalid " -"usernames will be rejected silently. Also, entering a username here " -"overrides the user's setting to not receive incoming calls. This way, you " -"can make certain users ring only for specific providers. Entries can be made " -"in a space-separated list, and/or one per line by hitting enter after every " -"one." -msgstr "" - -msgid "" -"For each user enabled for outgoing calls you can restrict what providers the " -"user can use for outgoing calls. By default all users can use all providers. " -"To show up in the list below the user should be allowed to make outgoing " -"calls in the \"User Accounts\" page. Enter VoIP providers in the format " -"username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest " -"to copy and paste the providers from above. Invalid entries, including " -"providers not enabled for outgoing calls, will be rejected silently. Entries " -"can be made in a space-separated list, and/or one per line by hitting enter " -"after every one." -msgstr "" - -msgid "Full Name" -msgstr "" - -msgid "General Settings" -msgstr "" - -msgid "Google Accounts" -msgstr "" - -msgid "Google Talk Status" -msgstr "" - -msgid "Google Talk Status Message" -msgstr "" - -msgid "Google Voice/Talk Accounts" -msgstr "" - -msgid "Hang-up Delay" -msgstr "" - -msgid "" -"Here you must configure at least one SIP account, that you will use to " -"register with this service. Use this account either in an Analog Telephony " -"Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid " -"on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By " -"default, all SIP accounts will ring simultaneously if a call is made to one " -"of your VoIP provider accounts or GV numbers." -msgstr "" - -msgid "" -"How long to wait before hanging up. If the provider you use to dial " -"automatically forwards to voicemail, you can set this value to a delay that " -"will allow you to hang up before your call gets forwarded and you get billed " -"for it." -msgstr "" - -msgid "" -"If setting Server/Registrar to %s or %s does not work for you, try setting " -"it to %s or %s and entering this port number in a separate field that " -"specifies the Server/Registrar port number. Beware that some devices have a " -"confusing setting that sets the port where SIP requests originate from on " -"the SIP device itself (the bind port). The port specified on this page is " -"NOT this bind port but the port this service listens on." -msgstr "" - -msgid "" -"If you experience jittery or high latency audio during heavy downloads, you " -"may want to enable QoS. QoS prioritizes traffic to and from your network for " -"specified ports and IP addresses, resulting in better latency and throughput " -"for sound in our case. If enabled below, a QoS rule for this service will be " -"configured by the PBX automatically, but you must visit the QoS " -"configuration page (Network->QoS) to configure other critical QoS settings " -"like Download and Upload speed." -msgstr "" - -msgid "" -"If you have more than one account that can make outgoing calls, you should " -"enter a list of phone numbers and/or prefixes in the following fields for " -"each provider listed. Invalid prefixes are removed silently, and only 0-9, " -"X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z " -"matches 1-9, and N matches 2-9. For example to make calls to Germany through " -"a provider, you can enter 49. To make calls to North America, you can enter " -"1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area " -"code like New York's 646, you can enter 646NXXXXXX for that provider. You " -"should leave one account with an empty list to make calls with it by " -"default, if no other provider's prefixes match. The system will " -"automatically replace an empty list with a message that the provider dials " -"all numbers not matched by another provider's prefixes. Be as specific as " -"possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international " -"dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a " -"space-separated list, and/or one per line by hitting enter after every one." -msgstr "" - -msgid "Incoming Calls" -msgstr "" - -msgid "Insert QoS Rules" -msgstr "" - -msgid "Makes Outgoing Calls" -msgstr "" - -msgid "NOTE: There are no Google or SIP provider accounts configured." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for incoming " -"calls." -msgstr "" - -msgid "" -"NOTE: There are no Google or SIP provider accounts enabled for outgoing " -"calls." -msgstr "" - -msgid "NOTE: There are no local user accounts configured." -msgstr "" - -msgid "NOTE: There are no local user accounts enabled for outgoing calls." -msgstr "" - -msgid "No" -msgstr "" - -msgid "Number of Seconds to Ring" -msgstr "" - -msgid "Outbound Proxy" -msgstr "" - -msgid "Outgoing Calls" -msgstr "" - -msgid "PBX Main Page" -msgstr "" - -msgid "PBX Service Status" -msgstr "" - -msgid "PIN" -msgstr "" - -msgid "Password" -msgstr "" - -msgid "" -"Pick a random port number between 6500 and 9500 for the service to listen " -"on. Do not pick the standard 5060, because it is often subject to brute-" -"force attacks. When finished, (1) click \"Save and Apply\", and (2) look in " -"the \"SIP Device/Softphone Accounts\" section for updated Server and Port " -"settings for your SIP Devices/Softphones." -msgstr "" - -msgid "Port Setting for SIP Devices" -msgstr "" - -msgid "Providers Used for Outgoing Calls" -msgstr "" - -msgid "QoS Settings" -msgstr "" - -msgid "RTP Port Range End" -msgstr "" - -msgid "RTP Port Range Start" -msgstr "" - -msgid "" -"RTP traffic carries actual voice packets. This is the start of the port " -"range that will be used for setting up RTP communication. It's usually OK to " -"leave this at the default value." -msgstr "" - -msgid "Receives Incoming Calls" -msgstr "" - -msgid "Remote Usage" -msgstr "" - -msgid "Rings users enabled for incoming calls" -msgstr "" - -msgid "SIP Accounts" -msgstr "" - -msgid "SIP Device/Softphone Accounts" -msgstr "" - -msgid "SIP Provider Accounts" -msgstr "" - -msgid "SIP Realm (needed by some providers)" -msgstr "" - -msgid "SIP Server/Registrar" -msgstr "" - -msgid "SIP Server/Registrar Port" -msgstr "" - -msgid "Server Setting" -msgstr "" - -msgid "Server Setting for Local SIP Devices" -msgstr "" - -msgid "Server Setting for Remote SIP Devices" -msgstr "" - -msgid "Service Status" -msgstr "" - -msgid "" -"Set the number of seconds to ring users upon incoming calls before hanging " -"up or going to voicemail, if the voicemail is installed and enabled." -msgstr "" - -msgid "Space-Separated List of Blacklisted Numbers" -msgstr "" - -msgid "Specify numbers individually here. Press enter to add more numbers." -msgstr "" - -msgid "" -"Specify numbers individually here. Press enter to add more numbers. You will " -"have to experiment with what country and area codes you need to add to the " -"number." -msgstr "" - -msgid "" -"The number(s) specified above will be able to dial out with this user's " -"providers. Invalid usernames, including users not enabled for outgoing " -"calls, are dropped silently. Please verify that the entry was accepted." -msgstr "" - -msgid "" -"This configuration page allows you to configure a phone system (PBX) service " -"which permits making phone calls through multiple Google and SIP (like " -"Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP " -"devices. Note that Google accounts, SIP accounts, and local user accounts " -"are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User " -"Accounts\" sub-sections. You must add at least one User Account to this PBX, " -"and then configure a SIP device or softphone to use the account, in order to " -"make and receive calls with your Google/SIP accounts. Configuring multiple " -"users will allow you to make free calls between all users, and share the " -"configured Google and SIP accounts. If you have more than one Google and SIP " -"accounts set up, you should probably configure how calls to and from them " -"are routed in the \"Call Routing\" page. If you're interested in using your " -"own PBX from anywhere in the world, then visit the \"Remote Usage\" section " -"in the \"Advanced Settings\" page." -msgstr "" - -msgid "" -"This is the name that the VoIP server will use to identify itself when " -"registering to VoIP (SIP) providers. Some providers require this to a " -"specific string matching a hardware SIP device." -msgstr "" - -msgid "" -"This is where you indicate which Google/SIP accounts are used to call what " -"country/area codes, which users can use what SIP/Google accounts, how " -"incoming calls are routed, what numbers can get into this PBX with a " -"password, and what numbers are blacklisted." -msgstr "" - -msgid "" -"This is where you set up your Google (Talk and Voice) Accounts, in order to " -"start using them for dialing and receiving calls (voice chat and real phone " -"calls). Please make at least one voice call using the Google Talk plugin " -"installable through the GMail interface, and then log out from your account " -"everywhere. Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This is where you set up your SIP (VoIP) accounts ts like Sipgate, " -"SipSorcery, the popular Betamax providers, and any other providers with SIP " -"settings in order to start using them for dialing and receiving calls (SIP " -"uri and real phone calls). Click \"Add\" to add as many accounts as you wish." -msgstr "" - -msgid "" -"This option should be set to \"Yes\" if you have a DID (real telephone " -"number) associated with this SIP account or want to receive SIP uri calls " -"through this provider." -msgstr "" - -msgid "" -"This section contains settings that do not need to be changed under normal " -"circumstances. In addition, here you can configure your system for use with " -"remote SIP devices, and resolve call quality issues by enabling the " -"insertion of QoS rules." -msgstr "" - -msgid "" -"Use (four to five digit) numeric user name if you are connecting normal " -"telephones with ATAs to this system (so they can dial user names)." -msgstr "" - -msgid "" -"Use this account to make outgoing calls as configured in the \"Call Routing" -"\" section." -msgstr "" - -msgid "Use this account to make outgoing calls." -msgstr "" - -msgid "User Accounts" -msgstr "" - -msgid "User Agent String" -msgstr "" - -msgid "User Name" -msgstr "" - -msgid "Uses providers enabled for outgoing calls" -msgstr "" - -msgid "" -"When somebody starts voice chat with your GTalk account or calls the GVoice, " -"number (if you have Google Voice), the call will be forwarded to any users " -"that are online (registered using a SIP device or softphone) and permitted " -"to receive the call. If you have Google Voice, you must go to your GVoice " -"settings and forward calls to Google chat in order to actually receive calls " -"made to your GVoice number. If you have trouble receiving calls from GVoice, " -"experiment with the Call Screening option in your GVoice Settings. Finally, " -"make sure no other client is online with this account (browser in gmail, " -"mobile/desktop Google Talk App) as it may interfere." -msgstr "" - -msgid "" -"When your password is saved, it disappears from this field and is not " -"displayed for your protection. The previously saved password will be changed " -"only when you enter a value different from the saved one." -msgstr "" - -msgid "Yes" -msgstr "" - -msgid "" -"You can enter your domain name, external IP address, or dynamic domain name " -"here. The best thing to input is a static IP address. If your IP address is " -"dynamic and it changes, your configuration will become invalid. Hence, it's " -"recommended to set up Dynamic DNS in this case. and enter your Dynamic DNS " -"hostname here. You can configure Dynamic DNS with the luci-app-ddns package." -msgstr "" - -msgid "You can specify a real name to show up in the Caller ID here." -msgstr "" - -msgid "" -"You can use your SIP devices/softphones with this system from a remote " -"location as well, as long as your Internet Service Provider gives you a " -"public IP. You will be able to call other local users for free (e.g. other " -"Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls " -"as if you were local to the PBX. After configuring this tab, go back to " -"where users are configured and see the new Server and Port setting you need " -"to configure the remote SIP devices with. Please note that if this PBX is " -"not running on your router/gateway, you will need to configure port " -"forwarding (NAT) on your router/gateway. Please forward the ports below (SIP " -"port and RTP range) to the IP address of the device running this PBX." -msgstr "" - -msgid "" -"Your PIN disappears when saved for your protection. It will be changed only " -"when you enter a value different from the saved one. Leaving the PIN empty " -"is possible, but please beware of the security implications." -msgstr "" - -msgid "" -"Your password disappears when saved for your protection. It will be changed " -"only when you enter a value different from the saved one." -msgstr "" |