diff options
author | Jo-Philipp Wich <jow@openwrt.org> | 2014-12-03 15:17:05 +0100 |
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committer | Jo-Philipp Wich <jow@openwrt.org> | 2015-01-08 16:26:20 +0100 |
commit | 1bb4822dca6113f73e3bc89e2acf15935e6f8e92 (patch) | |
tree | 35e16f100466e4e00657199b38bb3d87d52bf73f /applications/luci-app-pbx/po/el/pbx.po | |
parent | 9edd0e46c3f880727738ce8ca6ff1c8b85f99ef4 (diff) |
Rework LuCI build system
* Rename subdirectories to their repective OpenWrt package names
* Make each LuCI module its own standalone package
* Deploy a shared luci.mk which is used by each module Makefile
Signed-off-by: Jo-Philipp Wich <jow@openwrt.org>
Diffstat (limited to 'applications/luci-app-pbx/po/el/pbx.po')
-rw-r--r-- | applications/luci-app-pbx/po/el/pbx.po | 493 |
1 files changed, 493 insertions, 0 deletions
diff --git a/applications/luci-app-pbx/po/el/pbx.po b/applications/luci-app-pbx/po/el/pbx.po new file mode 100644 index 0000000000..717e2563b4 --- /dev/null +++ b/applications/luci-app-pbx/po/el/pbx.po @@ -0,0 +1,493 @@ +msgid "" +msgstr "" +"Project-Id-Version: PACKAGE VERSION\n" +"PO-Revision-Date: 2012-03-31 15:41+0200\n" +"Last-Translator: Vasilis <acinonyx@openwrt.gr>\n" +"Language-Team: none\n" +"Language: el\n" +"MIME-Version: 1.0\n" +"Content-Type: text/plain; charset=UTF-8\n" +"Content-Transfer-Encoding: 8bit\n" +"Plural-Forms: nplurals=2; plural=(n != 1);\n" +"X-Generator: Pootle 2.0.4\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 "Μην Ενοχλείτε" + +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 "Ενεργοποιημένο" + +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 "Πλήρες Όνομα" + +msgid "General Settings" +msgstr "Γενικές Ρυθμίσεις" + +msgid "Google Accounts" +msgstr "Λογαριασμοί Google" + +msgid "Google Talk Status" +msgstr "" + +msgid "Google Talk Status Message" +msgstr "" + +msgid "Google Voice/Talk Accounts" +msgstr "Λογαριασμοί Google Voice/Talk" + +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 "Εισερχόμενες Κλήσεις" + +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 "Όχι" + +msgid "Number of Seconds to Ring" +msgstr "" + +msgid "Outbound Proxy" +msgstr "" + +msgid "Outgoing Calls" +msgstr "Εξερχόμενες Κλήσεις" + +msgid "PBX Main Page" +msgstr "" + +msgid "PBX Service Status" +msgstr "" + +msgid "PIN" +msgstr "PIN" + +msgid "Password" +msgstr "Κωδικός πρόσβασης" + +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 "Λογαριασμοί SIP" + +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 "" + +#~ msgid "Account Status" +#~ msgstr "Κατάσταση Λογαριασμού" + +#~ msgid "Account Status Message" +#~ msgstr "Μήνυμα Κατάστασης Λογαριασμού" |