i18n: initialize swedish translations

1 files changed, 450 insertions, 0 deletions

diff --git a/po/sv/pbx.po b/po/sv/pbx.po
new file mode 100644
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+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"
+"Language: sv\n"
+"MIME-Version: 1.0\n"
+"Content-Transfer-Encoding: 8bit\n"
+"Plural-Forms: nplurals=2; plural=(n != 1);\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-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 it will have."
+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 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 ""
+"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 ""
+"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) click "
+"the \"Restart VoIP Service\" button above. Finally, (3) 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 ""
+"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 Please keep in mind that 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."
+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 ""