AddressCardFormTwinkle - Address CardTwinkle - Adresskort&Remark:Infix name of contact.First name of contact.Förnamn för kontakten.&First name:&Förnamn:You may place any remark about the contact here.&Phone:&Telefon:&Infix name:Phone number or SIP address of contact.Telefonnummer eller SIP-adress för kontakten.Last name of contact.Efternamn för kontakten.&Last name:&Efternamn:&OK&OKAlt+OAlt+O&Cancel&AvbrytAlt+CAlt+AYou must fill in a name.Du måste ange ett namn.You must fill in a phone number or SIP address.Du måste ange ett telefonnummer eller SIP-adress.AddressTableModelNameNamnPhoneTelefonRemarkAuthenticationFormTwinkle - AuthenticationTwinkle - AutentiseringuserNo need to translateanvändareThe user for which authentication is requested.Användaren för vilken autentisering begärs.profileNo need to translateprofilThe user profile of the user for which authentication is requested.Användarprofilen för användaren för vilken autentisering begärs.User profile:Användarprofil:User:Användare:&Password:&Lösenord:Your password for authentication.Ditt lösenord för autentisering.Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though.Ditt SIP-autentiseringsnamn. Ganska ofta är detta samma som ditt SIP-användarnamn. Det kan dock vara ett annat namn.&User name:Användar&namn:&OK&OK&Cancel&AvbrytLogin required for realm:realmNo need to translateThe realm for which you need to authenticate.BuddyFormTwinkle - BuddyTwinkle - KompisAddress bookAdressbokSelect an address from the address book.Välj en adress från adressboken.&Phone:&Telefon:Name of your buddy.Namnet på din kompis.&Show availability&Visa tillgänglighetAlt+SAlt+VCheck this option if you want to see the availability of your buddy. This will only work if your provider offers a presence agent.Kryssa för detta alternativ om du vill se tillgängligheten för din kompis. Detta kommer endast att fungera om din leverantör erbjuder en närvaroagent.&Name:&Namn:SIP address your buddy.SIP-adress till din kompis.&OK&OKAlt+OAlt+O&Cancel&AvbrytAlt+CAlt+AYou must fill in a name.Du måste ange ett namn.Invalid phone.Ogiltig telefon.Failed to save buddy list: %1Misslyckades med att spara kompislista: %1BuddyListAvailabilityTillgänglighetunknownokändofflinefrånkoppladonlineanslutenrequest failedbegäran misslyckadesrequest rejectedbegäran avvisadesnot publishedinte publiceradfailed to publishmisslyckades med att publiceraClick right to add a buddy.Högerklicka för att lägga till en kompis.CoreAudioFailed to open sound cardMIsslyckades med att öppna ljudkortetFailed to create a UDP socket (RTP) on port %1Misslyckades med att skapa ett UDP-uttag(RTP) på port %1Failed to create audio receiver thread.Misslyckades med att skapa ljudmottagartråd.Failed to create audio transmitter thread.Misslyckades med att skapa ljudsändartråd.CoreCallHistorylocal userlokal användareremote userfjärranvändarefailurefelunknownokändininoututDeregisterFormTwinkle - DeregisterTwinkle - Avregistreringderegister all devicesavregistrera alla enheter&OK&OK&Cancel&AvbrytDiamondcardProfileFormThis is just your full name, e.g. John Doe. It is used as a display name. When you make a call, this display name might be shown to the called party.Detta är helt enkelt ditt fullständiga namn, t.ex. Sven Svensson. Det används endast för visning. När du ringer ett samtal kan dock detta namn visas för motparten.&Your name:&Ditt namn:&OK&OK&Cancel&AvbrytFill in your account ID.Fill in your PIN code.A user profile with name %1 already exists.DtmfFormTwinkle - DTMFTwinkle - DTMFKeypadKnappsats2233Over decadic A. Normally not needed.Över dekadiskt A. Behövs oftast inte.445566Over decadic B. Normally not needed.Över dekadiskt B. Behövs oftast inte.778899Over decadic C. Normally not needed.Över dekadiskt C. Behövs oftast inte.Star (*)Stjärna (*)00Pound (#)Fyrkant (#)Over decadic D. Normally not needed.Över dekadiskt D. Behövs oftast inte.11&CloseS&tängAlt+CAlt+TFreeDeskSysTrayShow/HideVisa/GömQuitAvslutaGUIFailed to create a UDP socket (SIP) on port %1Kunde inte skapa UDP-socket (SIP) på port %1Override lock file and start anyway?Strunta i låsfil och starta ändå?The following profiles are both for user %1Följande profiler är båda för användaren %1You can only run multiple profiles for different users.Du kan endast köra flera profiler för olika användare.Cannot find a network interface. Twinkle will use 127.0.0.1 as the local IP address. When you connect to the network you have to restart Twinkle to use the correct IP address.Kan inte hitta ett nätverkskort. Twinkle kommer att använda 127.0.0.1 som lokal IP-adress. När du ansluter till nätverket så måste du starta om Twinkle för att använda den korrekta IP-adressen.Line %1: incoming call for %2Linje %1: inkommande samtal för %2Call transferred by %1Samtal kopplat av %1Line %1: far end cancelled call.Line %1: far end released call.Line %1: SDP answer from far end not supported.Linje %1: SDP-svar från andra parten stöds inte.Line %1: SDP answer from far end missing.Linje %1: Inget SDP-svar från andra parten.Line %1: Unsupported content type in answer from far end.Line %1: no ACK received, call will be terminated.Linje %1: inget ACK mottaget. Samtalet kommer avslutas.Line %1: no PRACK received, call will be terminated.Linje %1: inget PRACK mottaget. Samtalet kommer avslutas.Line %1: PRACK failed.Linje %1: PRACK misslyckades.Line %1: failed to cancel call.Linje %1: misslyckades med att avbryta samtal.Line %1: far end answered call.Linje %1: motparten svarade på samtalet.Line %1: call failed.Linje %1: samtal misslyckades.The call can be redirected to:Samtalet kan vidarekopplas till:Line %1: call released.Line %1: call established.Linje %1: samtal etablerat.Response on terminal capability request: %1 %2Svar på begäran om terminalförmågor: %1 %2Terminal capabilities of %1Terminalförmågor för %1Accepted body types:unknownokäntAccepted encodings:Accepterade kodningar:Accepted languages:Accepterade språk:Allowed requests:Tillåtna begäran:Supported extensions:noneEnd point type:Typ av motpart:Line %1: call retrieve failed.%1, registration failed: %2 %3%1, registrering misslyckades: %2 %3%1, registration succeeded (expires = %2 seconds)%1, registrering lyckades (går ut om = %2 sekunder)%1, registration failed: STUN failure%1, registrering misslyckades: STUN-problem%1, de-registration succeeded: %2 %3%1, avregistrering lyckades: %2 %3%1, de-registration failed: %2 %3%1, avregistrering misslyckades: %2 %3%1, fetching registrations failed: %2 %3: you are not registered: du är inte registrerad: you have the following registrations: du har följande registreringar: fetching registrations...: hämtar registreringar...Line %1: redirecting request toLinje %1: Skickar vidare tillRedirecting request to: %1Skickar vidare till: %1Line %1: DTMF detected:Linje %1: DTMF detekterad:invalid DTMF telephone event (%1)Line %1: send DTMF %2Linje %1: skicka DTMF %2Line %1: far end does not support DTMF telephone events.Line %1: received notification.Event: %1Händelse: %1State: %1Tillstånd: %1Reason: %1Anledning: %1Progress: %1 %2Förlopp: %1 %2Line %1: call transfer failed.Linje %1: samtalskoppling misslyckades.Line %1: call successfully transferred.Linje %1: samtal kopplades.Line %1: call transfer still in progress.Linje %1: samtalskoppling pågår fortfarande.No further notifications will be received.Line %1: transferring call to %2Linje %1: kopplar samtal till %2Transfer requested by %1Koppilng begärd av %1Line %1: Call transfer failed. Retrieving original call.%1, STUN request failed: %2 %3%1. STUN-begäran misslyckades: %2 %3%1, STUN request failed.%1, STUN-begäran misslyckades.Redirecting callVidarekoppling av samtalUser profile:Användarprofil:User:Användare:Do you allow the call to be redirected to the following destination?Tillåter du samtalet att hänvisas till följande destination?If you don't want to be asked this anymore, then you must change the settings in the SIP protocol section of the user profile.Om du inte vill bli frågad detta igen måste du ändra inställningarna i sektionen SIP-protokoll för användarprofilen.Redirecting requestDo you allow the %1 request to be redirected to the following destination?Transferring callRequest to transfer call received from:Request to transfer call received.Do you allow the call to be transferred to the following destination?Tillåter du samtalet att kopplas vidare till följande destination?Info:Warning:Varning:Critical:Firewall / NAT discovery...Brandvägg / NAT-identifiering...AbortAvbrytLine %1Linje %1Click the padlock to confirm a correct SAS.Klicka på hänglåset för att bekräfta en korrekt SAS.The remote user on line %1 disabled the encryption.Motparten på linje %1 stängde av krypteringen.Line %1: SAS confirmed.Linje %1: SAS bekräftad.Line %1: SAS confirmation reset.%1, voice mail status failure.%1, kan inte hämta status för röstbrevlåda.%1, voice mail status rejected.%1, hämtning av status för röstbrevlåda avvisades.%1, voice mailbox does not exist.%1, röstbrevlådan finns inte.%1, voice mail status terminated.%1, hämtning av status för röstbrevlåda avslutad.Line %1: call rejected.Linje %1: samtal avvisades.Line %1: call redirected.Linje %1: samtal vidarekopplat.Failed to start conference.Kunde inte starta konferens.Failed to create a %1 socket (SIP) on port %2Misslyckades med att skapa ett %1-uttag (SIP) på port %2If these are users for different domains, then enable the following option in your user profile (SIP protocol)Use domain name to create a unique contact headerFailed to save message attachment: %1Misslyckades med att spara meddelandebilaga: %1Accepted by networkAccepterad av nätverketTransferred by: %1Cannot open web browser: %1Configure your web browser in the system settings.GetAddressFormTwinkle - Select addressTwinkle - Välj adress&KAddressBook&KAddressBookNameNamnTypeTypPhoneTelefonThis list of addresses is taken from <b>KAddressBook</b>. Contacts for which you did not provide a phone number are not shown here. To add, delete or modify address information you have to use KAddressBook.&Show only SIP addresses&Visa endast SIP-adresserAlt+SAlt+VCheck this option when you only want to see contacts with SIP addresses, i.e. starting with "<b>sip:</b>".Kryssa för detta alternativ när du endast vill se kontakter med SIP-adresser, alltså börjar med "<b>sip:</b>".&ReloadUpp&dateraAlt+RAlt+DReload the list of addresses from KAddressbook.Uppdatera listan över adresser från KAddressbook.&Local address book&Lokal adressbokContacts in the local address book of Twinkle.Kontakter i den lokala adressboken för Twinkle.&Add&Lägg tillAlt+AAlt+LAdd a new contact to the local address book.Lägg till en ny kontakt i den lokala adressboken.&Delete&Ta bortAlt+DAlt+TDelete a contact from the local address book.Ta bort en kontakt från den lokala adressboken.&Edit&RedigeraAlt+EAlt+REdit a contact from the local address book.Redigera en kontakt från den lokala adressboken.&OK&OKAlt+OAlt+O&Cancel&AvbrytAlt+CAlt+A<p>You seem not to have any contacts with a phone number in <b>KAddressBook</b>, KDE's address book application. Twinkle retrieves all contacts with a phone number from KAddressBook. To manage your contacts you have to use KAddressBook.<p>As an alternative you may use Twinkle's local address book.</p>Are you sure you want to delete contact '%1' from the local address book?Delete contactGetProfileNameFormTwinkle - Profile nameTwinkle - Profilnamn&OK&OK&Cancel&AvbrytEnter a name for your profile:Ge din profil ett namn:<b>The name of your profile</b>
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A profile contains your user settings, e.g. your user name and password. You have to give each profile a name.
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If you have multiple SIP accounts, you can create multiple profiles. When you startup Twinkle it will show you the list of profile names from which you can select the profile you want to run.
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To remember your profiles easily you could use your SIP user name as a profile name, e.g. <b>example@example.com</b><b>Din profils namn</b>
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En profil innehåller dina användarinställningar. T.ex. ditt användarnamn och lösenord. Du måste ge varje profil ett namn.
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Om du har flera SIP-konton kan du skapa flera profiler. När du startar Twinkle kommer du se en lista över dina profiler och kunna välja vilken av profilerna du vill använda.
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För att enkelt komma ihåg dina profiler kan du använda ditt SIP-användarnamn som profilnamn. T.ex. <b>exempel@exempel.com</b>Cannot find .twinkle directory in your home directory.Kan inte hitta katalogen .twinkle i din hemkatalog.Profile already exists.Profilen finns redan.Rename profile '%1' to:Döp om profilen '%1' till:HistoryFormTwinkle - Call HistoryTwinkle - SamtalshistorikTimeTidIn/OutIn/UtFrom/ToFrån/TillSubjectÄmneStatusStatusCall detailsSamtalsdetaljerDetails of the selected call record.Detaljer för det valda samtalet.ViewVisa&Incoming calls&Inkommande samtalAlt+IAlt+ICheck this option to show incoming calls.Kryssa i denna ruta för att visa inkommande samtal.&Outgoing calls&Utgående samtalAlt+OAlt+UCheck this option to show outgoing calls.Kryssa i denna ruta för att visa utgående samtal.&Answered calls&Besvarade samtalAlt+AAlt+BCheck this option to show answered calls.Kryssa i denna ruta för att visa besvarade samtal.&Missed calls&Missade samtalAlt+MAlt+MCheck this option to show missed calls.Kryssa i denna ruta för att visa missade samtal.Current &user profiles onlyEnbart aktuell &användarprofilAlt+UAlt+ACheck this option to show only calls associated with this user profile.Kryssa i denna ruta för att enbart visa samtal associerade med aktuell användarprofil.C&lear&TömAlt+LAlt+T<p>Clear the complete call history.</p>
<p><b>Note:</b> this will clear <b>all</b> records, also records not shown depending on the checked view options.</p><p>Töm hela samtalshistoriken.</p>
<p><b>Observera:</b> detta rensar bort <b>alla</b> samtal. Även samtal som inte visas på grund av aktuella visningsalternativ.</p>&Close&StängAlt+CAlt+SClose this window.Stäng detta fönster.Call start:Samtal startade:Call answer:Samtal besvarades:Call end:Samtal avslutades:Call duration:Samtalslängd:Direction:Riktning:From:Från:To:Till:Reply to:Svara till:Referred by:Kopplad av:Subject:Ämne:Released by:Avslutat av:Status:Status:Far end device:Enhet på andra sidan:User profile:Användarprofil:conversationkonversationCall...Ring upp...DeleteTa bortRe:Sv:Clo&seSt&ängAlt+SAlt+Ä&Call&RingCall selected address.Ring markerad adress.Number of calls:###Total call duration:IncomingCallPopup%1 callingInviteFormTwinkle - CallTwinkle - Ring&To:&Till:Optionally you can provide a subject here. This might be shown to the callee.Om du vill kan du ange ämne här. Det kan visas för den du ringer.Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.The address that you want to call. This can be a full SIP address like <b>sip:example@example.com</b> or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile.Adressen som du vill ringa. Detta kan vara en komplett SIP-adress som <b>sip:exempel@exempel.com</b> eller bara användarnamnet eller telefonnumret i adressen. När du inte anger en komplett adress kommer Twinkle fylla ut adressen med domännamnet från din användarprofil.The user that will make the call.Användare som ringer upp.&Subject:&Ämne:&From:&Från:&Hide identity&Dölj identitetAlt+HAlt+D<p>
With this option you request your SIP provider to hide your identity from the called party. This will only hide your identity, e.g. your SIP address, telephone number. It does <b>not</b> hide your IP address.
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<p>
<b>Warning:</b> not all providers support identity hiding.
</p><p>
Med denna inställning kan du be din SIP-leverantör dölja din identitet för den du ringer upp. Detta döljer bara din identitet. D.v.s. din SIP-adress eller telefonnummer. Det döljer <b>inte</b> din IP-adress.
</p>
<p>
<b>Varning:</b> Inte alla SIP-leverantörer stöder dold identitet.
</p>&OK&OK&Cancel&AvbrytNot all SIP providers support identity hiding. Make sure your SIP provider supports it if you really need it.Inte alla SIP-leverantörer stöder dold identitet. Se till att din SIP-leverantör stöder det om du verkligen behöver det.F10F10LogViewFormTwinkle - LogTwinkle - LoggContents of the current log file (~/.twinkle/twinkle.log)Innehåll i nuvarande loggfil (~/.twinkle/twinkle.log)&Close&StängAlt+CAlt+SC&lear&TömAlt+LAlt+TClear the log window. This does <b>not</b> clear the log file itself.Töm loggfönstret. Detta tömmer <b>inte</b> själva loggfilen.MessageFormTwinkle - Instant messageTwinkle - Snabbmeddelande&To:&Till:The user that will send the message.Användaren som ska skicka meddelandet.The address of the user that you want to send a message. This can be a full SIP address like <b>sip:example@example.com</b> or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile.Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.&User profile:&Användarprofil:ConversationKonversationType your message here and then press "send" to send it.Skriv ditt meddelande här och tryck sedan på "Skicka" för att skicka det.&Send&SkickaAlt+SAlt+SSend the message.Skicka meddelandet.Instant message toolbarVerktygsrad för snabbmeddelandenSend file...Skicka fil...Send fileSkicka filimage size is scaled down in previewbildstorleken är nedskalad i förhandsvisningDelivery failureLeverans misslyckadesDelivery notificationLeveransnotifieringOpen with %1...Öppna med %1...Open with...Öppna med...Save attachment as...Spara bilaga som...File already exists. Do you want to overwrite this file?Filen finns redan. Vill du skriva över denna fil?Failed to save attachment.Misslyckades med att spara bilaga.%1 is typing a message.%1 skriver ett meddelande.F10F10SizeMessageFormViewsending messageskickar meddelandeMphoneFormTwinkleTwinkle&Call:Label in front of combobox to enter address&Ring:The address that you want to call. This can be a full SIP address like <b>sip:example@example.com</b> or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile.Adressen som du vill ringa. Detta kan vara en komplett SIP-adress som <b>sip:exempel@exempel.com</b> eller bara användarnamnet eller telefonnumret i adressen. När du inte anger en komplett adress kommer Twinkle fylla ut adressen med domännamnet från din användarprofil.Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.DialRingDial the address.Ring till adressen.&User:&Användare:The user that will make the call.Användaren som ska ringa samtalet.Auto answer indication.Call redirect indication.Do not disturb indication.Missed call indication.Registration status.DisplayLine statusLinjestatusLine &1:Linje &1:Alt+1Alt+1Click to switch to line 1.Klicka för att växla till linje 1.From:Från:To:Till:Subject:Ämne:idleNo need to translateOverksamConference callKonferenssamtalTransferring callsasNo need to translatesasShort authentication stringg711a/g711aNo need to translateg711a/g711aAudio codecLjudkodek0:00:000:00:00Call durationSamtalslängdsip:fromNo need to translatesip:frånsip:toNo need to translatesip:tillsubjectNo need to translateämnephotoNo need to translatefotoLine &2:Linje &2:Alt+2Alt+2Click to switch to line 2.Klicka för att växla till linje 2.&File&Arkiv&Edit&RedigeraC&all&RingActivate lineAktivera linje&Registration&Registrering&Services&Tjänster&View&Visa&Help&HjälpQuitAvsluta&Quit&AvslutaCtrl+QCtrl+AAbout TwinkleOm Twinkle&About Twinkle&Om TwinkleCalltoolbar textRing&Call...call menu text&Ring...Call someoneRing någonF5F5Answertoolbar textSvara&Answermenu text&SvaraAnswer incoming callSvara inkommande samtalF6F6Byetoolbar textAdjö&Byemenu text&Lägg påRelease callEscEscRejecttoolbar textAvvisa&Rejectmenu text&AvvisaReject incoming callAvvisa inkommande samtalF8F8&Holdmenu text&ParkeraPut a call on hold, or retrieve a held callPakera ett samtal eller återuppta ett parkerat samtalRedirecttoolbar textKoppla vidareR&edirect...menu textKoppla vidar&e...Redirect incoming call without answeringKoppla vidare inkommande samtal utan att svaraDtmftoolbar textDtmf&Dtmf...menu text&Dtmf...Open keypad to enter digits for voice menu'sÖppna knappsats för att mata in siffror för röstmenyerRegisterRegistrera&Register&RegistreraDeregisterAvregistrera&DeregisterAvre&gistreraDeregister this deviceAvregistrera denna enhetShow registrationsVisa registreringar&Show registrations&Visa registreringarTerminal capabilitiesTerminalförmågor&Terminal capabilities...menu text&Terminalförmågor...Request terminal capabilities from someoneBegär terminalförmågor från någonDo not disturbStör inte&Do not disturbStör &inteCall redirectionVidarekoppla samtalCall &redirection...Vidarekoppla sa&mtal...Redialtoolbar textRing igen&Redialmenu textRing &igenRepeat last callUpprepa senaste samtaletF12F12About QtOm QtAbout &QtOm &QtUser profileAnvändarprofil&User profile...&Användarprofil...Conftoolbar textKonf&Conferencemenu text&KonferensJoin two calls in a 3-way conferenceMutetoolbar textTyst&Mutemenu text&TystMute a callStäng av mikrofonenXfertoolbar textKopplaTrans&fer...menu textKopp&la...Transfer callKoppla samtalSystem settingsSysteminställningar&System settings...&Systeminställningar...Deregister allAvregistrera allaDeregister &allAvregistrera &allaDeregister all your registered devicesAvregistera alla dina registrerade enheterAuto answerSvara automatiskt&Auto answerS&vara automatisktLogLogg&Log...&Logg...Call historySamtalshistorikCall &history...Samtals&historik...F9F9Change user ...Byt användare...&Change user ...&Byt användare...Activate or de-activate usersAktivera eller inaktivera användareWhat's This?Vad är detta?What's &This?Vad är &detta?Shift+F1Shift+F1Line 1Linje 1Line 2Linje 2idleoverksamdialingringerattempting call, please waitförsöker ringa samtal, väntaincoming callinkommande samtalestablishing call, please waitetablerar samtal, väntaestablishedetableratestablished (waiting for media)etablerat (väntar på media)releasing call, please waitunknown stateokänt tillståndVoice is encryptedRöstkanal är krypteradClick to confirm SAS.Klicka för att bekräfta SAS.Click to clear SAS verification.Transfer consultationUser:Användare:Call:Ring:Hide identityDölj identitetRegistration status:Registreringsstatus:RegisteredRegistreradFailedMisslyckadesNot registeredInte registreradClick to show registrations.Klicka för att visa registreringar.No users are registered.Inga användare är registrerade.%1 new, 1 old message%1 nytt, ett gammalt meddelande%1 new, %2 old messages%1 nya, %2 gamla meddelanden1 new messageEtt nytt meddelande%1 new messages%1 nya meddelanden1 old messageEtt gammalt meddelande%1 old messages%1 gamla meddelandenMessages waitingMeddelanden väntarNo messagesInga meddelanden<b>Voice mail status:</b><b>Status för röstbrevlåda:</b>FailureMisslyckadesUnknownOkäntClick to access voice mail.Klicka för att komma åt röstmeddelanden.Do not disturb active for:Stör inte är aktivt för:Redirection active for:Vidarekoppling är aktivt för:Auto answer active for:Svara automatiskt är aktivt för:Click to activate/deactivateKlicka för att aktivera/inaktiveraClick to activateKlicka för att aktiveraDo not disturb is not active.Stör inte är inte aktiverat.Redirection is not active.Vidarekoppling är inte aktiverat.Auto answer is not active.Svara automatiskt är inte aktiverat.Click to see call history for details.Klicka för att se samtalshistorik för detaljer.You have no missed calls.Du har inga missade samtal.You missed 1 call.Du missade ett samtal.You missed %1 calls.Du missade %1 samtal.Starting user profiles...Startar användarprofiler...The following profiles are both for user %1Följande profiler är båda för användaren %1You can only run multiple profiles for different users.Du kan endast köra flera profiler för olika användare.You have changed the SIP UDP port. This setting will only become active when you restart Twinkle.not provisionedYou must provision your voice mail address in your user profile, before you can access it.The line is busy. Cannot access voice mail.Linjen är upptagen. Kan inte komma åt röstmeddelanden.The voice mail address %1 is an invalid address. Please provision a valid address in your user profile.Buddy listKompislistaYou can create a separate buddy list for each user profile. You can only see availability of your buddies and publish your own availability if your provider offers a presence server.Message waiting indication.&Message&Meddelande&DisplayVoice mailRöstbrevlåda&Voice mail&RöstbrevlådaAccess voice mailF11F11MsgChattInstant &message...Snabb&meddelande...Instant messageSnabbmeddelande&Buddy list&Kompislista&Call...&Ring...&Edit...&Redigera...&Delete&Ta bortO&fflineFr&ånkopplad&OnlineA&nsluten&Change availability&Ändra tillgänglighet&Add buddy...&Lägg till kompis...Failed to save buddy list: %1Misslyckades med att spara kompislista. %1F10F10DiamondcardManual&ManualSign up&Sign up...Recharge...Balance history...Call history...Admin center...RechargeBalance historyAdmin centerCallRing&Answer&SvaraAnswerSvara&Bye&Lägg påByeAdjö&Reject&AvvisaRejectAvvisa&Hold&ParkeraHoldR&edirect...Koppla vidar&e...RedirectKoppla vidare&Dtmf...&Dtmf...DtmfDtmf&Terminal capabilities...&Terminalförmågor...&RedialRing &igenRedialRing igen&Conference&KonferensConfKonf&Mute&TystMuteTystTrans&fer...Kopp&la...XferKopplaNumberConversionFormTwinkle - Number conversionTwinkle - Nummerkonvertering&Match expression:&Matcha uttryck:&Replace:&Ersätt:Perl style format string for the replacement number.Perl style regular expression matching the number format you want to modify.&OK&OKAlt+OAlt+O&Cancel&AvbrytAlt+CAlt+AMatch expression may not be empty.Uttryck att matcha mot kan inte vara tomt.Replace value may not be empty.Ersättningsvärde kan inte vara tomt.Invalid regular expression.Ogiltigt reguljärt uttryck.RedirectFormTwinkle - RedirectTwinkle - Koppla vidareRedirect incoming call toKoppla vidare inkommande samtal tillYou can specify up to 3 destinations to which you want to redirect the call. If the first destination does not answer the call, the second destination will be tried and so on.Du kan ange upp till 3 destinationer att koppla vidare samtalet till. Om första destinationen inte svarar går man vidare till andra destinationen och så vidare.&3rd choice destination:Destination i &tredje hand:&2nd choice destination:Destination i &andra hand:&1st choice destination:Destination i &första hand:Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.&OK&OK&Cancel&AvbrytF10F10F12F12F11F11SelectNicFormTwinkle - Select NICTwinkle - Välj nätverkskortSelect the network interface/IP address that you want to use:Välj nätverkskort/IP-adress som du vill använda:You have multiple IP addresses. Here you must select which IP address should be used. This IP address will be used inside the SIP messages.Du har flera IP-adresser. Här måste du välja vilken IP-adress som ska användas. Denna IP-adress kommer användas i alla SIP-meddelanden.Set as default &IPAnge som standard-&IPAlt+IAlt+IMake the selected IP address the default IP address. The next time you start Twinkle, this IP address will be automatically selected.Gör den valda IP-adressen till standard-IP-adress. Nästa gång som du startar Twinkle kommer denna IP-adress att väljas automatiskt.Set as default &NICAnge som standard&nätverkskortAlt+NAlt+NMake the selected network interface the default interface. The next time you start Twinkle, this interface will be automatically selected.Gör det valda nätverkskortet till standardgränssnitt. Nästa gång som du startar Twinkle kommer detta nätverkskort att väljas automatiskt.&OK&OKAlt+OAlt+OIf you want to remove or change the default at a later time, you can do that via the system settings.Om du vill ta bort eller ändra standardvärdet vid ett senare tillfälle så kan du göra det via systeminställningarna.SelectProfileFormTwinkle - Select user profileTwinkle - Välj användarprofilSelect user profile(s) to run:Välj användarprofil(er) att köra:User profileAnvändarprofilTick the check boxes of the user profiles that you want to run and press run.Kryssa för de användarprofiler som du vill köra och tryck på Kör.&New&NyAlt+NAlt+NCreate a new profile with the profile editor.Skapa en ny profil med profilredigeraren.&Wizard&GuideAlt+WAlt+GCreate a new profile with the wizard.Skapa en ny profil med guiden.&Edit&RedigeraAlt+EAlt+REdit the highlighted profile.Redigera markerad profil.&Delete&Ta bortAlt+DAlt+TDelete the highlighted profile.Ta bort markerad profil.Ren&ameByt &namnAlt+AAlt+NRename the highlighted profile.Byt namn på markerad profil.&Set as default&Ange som standardAlt+SAlt+AMake the selected profiles the default profiles. The next time you start Twinkle, these profiles will be automatically run.Gör de markerade profilerna till standardprofiler. Nästa gång som du startar Twinkle så kommer dessa profiler att köras automatiskt.&Run&KörAlt+RAlt+KRun Twinkle with the selected profiles.Kör Twinkle med markerade profiler.S&ystem settingsS&ysteminställningarAlt+YAlt+YEdit the system settings.Redigera systeminställningarna.&Cancel&AvbrytAlt+CAlt+A<html>Before you can use Twinkle, you must create a user profile.<br>Click OK to create a profile.</html><html>Innan du kan använda Twinkle måste du skapa en användarprofil.<br>Klicka OK för att skapa en profil.</html>&Profile editor&Profilredigerare<html>Next you may adjust the system settings. You can change these settings always at a later time.<br><br>Click OK to view and adjust the system settings.</html>You did not select any user profile to run.
Please select a profile.Du valde inte någon användarprofil att köra.
Välj en profil.Are you sure you want to delete profile '%1'?Är du säker på att du vill ta bort profilen "%1"?Delete profileTa bort profilFailed to delete profile.Misslyckades med att ta bort profil.Failed to rename profile.Misslyckades med att byta namn på profil.<p>If you want to remove or change the default at a later time, you can do that via the system settings.</p>Cannot find .twinkle directory in your home directory.Kan inte hitta katalogen .twinkle i din hemkatalog.Create profileEd&itorAlt+IAlt+IDia&mondcardAlt+MAlt+MModify profileStartup profile&DiamondcardCreate a profile for a Diamondcard account. With a Diamondcard account you can make worldwide calls to regular and cell phones and send SMS messages.<html>You can use the profile editor to create a profile. With the profile editor you can change many settings to tune the SIP protocol, RTP and many other things.<br><br>Alternatively you can use the wizard to quickly setup a user profile. The wizard asks you only a few essential settings. If you create a user profile with the wizard you can still edit the full profile with the profile editor at a later time.<br><br>You can create a Diamondcard account to make worldwide calls to regular and cell phones and send SMS messages.<br><br>Choose what method you wish to use.</html>SelectUserFormTwinkle - Select userTwinkle - Välj användare&Cancel&AvbrytAlt+CAlt+A&Select all&Välj allaAlt+SAlt+V&OK&OKAlt+OAlt+OC&lear all&Töm allaAlt+LAlt+TpurposeNo need to translatesyfteUserAnvändareRegisterRegistreraSelect users that you want to register.Välj användare som du vill registrera.DeregisterAvregistreraSelect users that you want to deregister.Välj användare som du vill avregistrera.Deregister all devicesAvregistrera alla enheterSelect users for which you want to deregister all devices.Välj användare för vilka du vill avregistrera alla enheter.Do not disturbStör inteSelect users for which you want to enable 'do not disturb'.Välj användare för vilka du vill aktivera 'stör ej'.Auto answerSvara automatisktSelect users for which you want to enable 'auto answer'.Välj användare för vilka du vill aktivera "svara automatiskt".SendFileFormTwinkle - Send FileTwinkle - Skicka filSelect file to send.Välj fil att skicka.&File:&Fil:&Subject:&Ämne:&OK&OKAlt+OAlt+O&Cancel&AvbrytAlt+CAlt+AFile does not exist.Filen finns inte.Send file...Skicka fil...SrvRedirectFormTwinkle - Call RedirectionUser:Användare:There are 3 redirect services:<p>
<b>Unconditional:</b> redirect all calls
</p>
<p>
<b>Busy:</b> redirect a call if both lines are busy
</p>
<p>
<b>No answer:</b> redirect a call when the no-answer timer expires
</p>&Unconditional&Redirect all callsAlt+RActivate the unconditional redirection service.Redirect toYou can specify up to 3 destinations to which you want to redirect the call. If the first destination does not answer the call, the second destination will be tried and so on.Du kan ange upp till 3 destinationer att koppla vidare samtalet till. Om första destinationen inte svarar går man vidare till andra destinationen och så vidare.&3rd choice destination:Destination i &3:e hand:&2nd choice destination:Destination i &2:a hand:&1st choice destination:Destination i &1:a hand:Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.&Busy&Upptagen&Redirect calls when I am busyActivate the redirection when busy service.&No answer&Inget svar&Redirect calls when I do not answerActivate the redirection on no answer service.&OK&OKAlt+OAlt+OAccept and save all changes.Acceptera och spara alla ändringar.&Cancel&AvbrytAlt+CAlt+AUndo your changes and close the window.Ångra dina ändringar och stäng fönstret.You have entered an invalid destination.Du har angivit en ogiltig destination.F10F10F11F11F12F12SysSettingsFormTwinkle - System SettingsTwinkle - SysteminställningarGeneralGenerelltAudioLjudRing tonesRingsignalerAddress bookAdressbokNetworkNätverkLogLoggSelect a category for which you want to see or modify the settings.Välj den kategori som du vill titta eller ändra inställningar för.&OK&OKAlt+OAlt+OAccept and save your changes.Acceptera och spara dina ändringar.&Cancel&AvbrytAlt+CAlt+AUndo all your changes and close the window.Ångra alla ändringar och stäng fönstret.Sound CardLjudkortSelect the sound card for playing the ring tone for incoming calls.Välj det ljudkort som ska användas för att spela upp ringsignalen när någon ringer.Select the sound card to which your microphone is connected.Välj det ljudkort som din mikrofon är inkopplad till.Select the sound card for the speaker function during a call.Välj det ljudkort som högtalaren är inkopplad till under samtal.&Speaker:&Högtalare:&Ring tone:&Ringsignal:Other device:Annan enhet:&Microphone:&Mikrofon:&Validate devices before usage&Validera enheterna innan användningAlt+VAlt+V<p>
Twinkle validates the audio devices before usage to avoid an established call without an audio channel.
<p>
On startup of Twinkle a warning is given if an audio device is inaccessible.
<p>
If before making a call, the microphone or speaker appears to be invalid, a warning is given and no call can be made.
<p>
If before answering a call, the microphone or speaker appears to be invalid, a warning is given and the call will not be answered.Reduce &noise from the microphoneReducera &brus från mikrofonenAlt+NAlt+BAdvancedAvanceratOSS &fragment size:161632326464128128256256The ALSA play period size influences the real time behaviour of your soundcard for playing sound. If your sound frequently drops while using ALSA, you might try a different value here.ALSA &play period size:&ALSA capture period size:The OSS fragment size influences the real time behaviour of your soundcard. If your sound frequently drops while using OSS, you might try a different value here.The ALSA capture period size influences the real time behaviour of your soundcard for capturing sound. If the other side of your call complains about frequently dropping sound, you might try a different value here.&Max log size:&Maximal loggstorlek:The maximum size of a log file in MB. When the log file exceeds this size, a backup of the log file is created and the current log file is zapped. Only one backup log file will be kept.MBMBLog &debug reportsAlt+DAlt+TIndicates if reports marked as "debug" will be logged.Log &SIP reportsLogga &SIP-rapporterAlt+SAlt+SIndicates if SIP messages will be logged.Log S&TUN reportsLogga S&TUN-rapporterAlt+TAlt+TIndicates if STUN messages will be logged.Log m&emory reportsAlt+EAlt+RIndicates if reports concerning memory management will be logged.System trayAktivitetsfältCreate &system tray icon on startupSkapa ikon i a&ktivitetsfält vid uppstartEnable this option if you want a system tray icon for Twinkle. The system tray icon is created when you start Twinkle.&Hide in system tray when closing main windowAlt+HAlt+DEnable this option if you want Twinkle to hide in the system tray when you close the main window.StartupUppstartS&tartup hidden in system trayNext time you start Twinkle it will immediately hide in the system tray. This works best when you also select a default user profile.Default user profilesAnvändarprofiler som standardIf you always use the same profile(s), then you can mark these profiles as default here. The next time you start Twinkle, you will not be asked to select which profiles to run. The default profiles will automatically run.ServicesTjänsterCall &waitingSamtal &väntarAlt+WAlt+VWith call waiting an incoming call is accepted when only one line is busy. When you disable call waiting an incoming call will be rejected when one line is busy.Hang up &both lines when ending a 3-way conference call.Alt+BAlt+BHang up both lines when you press bye to end a 3-way conference call. When this option is disabled, only the active line will be hung up and you can continue talking with the party on the other line.&Maximum calls in call history:The maximum number of calls that will be kept in the call history.&Auto show main window on incoming call afterAlt+AAlt+BWhen the main window is hidden, it will be automatically shown on an incoming call after the number of specified seconds.Number of seconds after which the main window should be shown.secss&RTP port:&RTP-port:The UDP port used for sending and receiving RTP for the first line. The UDP port for the second line is 2 higher. E.g. if port 8000 is used for the first line, then the second line uses port 8002. When you use call transfer then the next even port (eg. 8004) is also used.Ring toneRingsignal&Play ring tone on incoming callS&pela upp ringsignal vid inkommande samtalAlt+PAlt+PIndicates if a ring tone should be played when a call comes in.Indikerar om en ringsignal ska spelas upp när ett samtal kommer in.&Default ring toneStan&dardringsignalPlay the default ring tone when a call comes in.Spela upp standardringsignalen när ett samtal kommer in.C&ustom ring toneA&npassad ringsignalAlt+UAlt+NPlay a custom ring tone when a call comes in.Spela upp en anpassad ringsignal när ett samtal kommer in.Specify the file name of a .wav file that you want to be played as ring tone.Ring back toneP&lay ring back tone when network does not play ring back toneAlt+LAlt+T<p>
Play ring back tone while you are waiting for the far-end to answer your call.
</p>
<p>
Depending on your SIP provider the network might provide ring back tone or an announcement.
</p>D&efault ring back tonePlay the default ring back tone.Cu&stom ring back tonePlay a custom ring back tone.Specify the file name of a .wav file that you want to be played as ring back tone.&Lookup name for incoming callOn an incoming call, Twinkle will try to find the name belonging to the incoming SIP address in your address book. This name will be displayed.Ove&rride received display nameAlt+RThe caller may have provided a display name already. Tick this box if you want to override that name with the name you have in your address book.Lookup &photo for incoming callSlå upp &foto för inkommande samtalLookup the photo of a caller in your address book and display it on an incoming call.noneThis is the 'none' in default IP address comboingennoneThis is the 'none' in default network interface comboingenRing tonesDescription of .wav files in file dialogRingsignalerChoose ring toneVälj ringsignalRing back tonesDescription of .wav files in file dialogChoose ring back toneMaximum allowed size (0-65535) in bytes of an incoming SIP message over UDP.&SIP port:&SIP-port:Max. SIP message size (&TCP):The UDP/TCP port used for sending and receiving SIP messages.Max. SIP message size (&UDP):Maximum allowed size (0-4294967295) in bytes of an incoming SIP message over TCP.Select ring tone file.Select ring back tone file.W&eb browser command:Command to start your web browser. If you leave this field empty Twinkle will try to figure out your default web browser.51251210241024Tip: for crackling sound with PulseAudio, set play period size to maximum.Enable in-call OSDSysTrayPopupAnswerSvaraRejectAvvisaIncoming CallTermCapFormTwinkle - Terminal CapabilitiesTwinkle - Terminalförmågor&From:&Från:Get terminal capabilities ofHämta terminalförmågor för&To:&Till:The address that you want to query for capabilities (OPTION request). This can be a full SIP address like <b>sip:example@example.com</b> or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile.Adressen som du vill fråga efter förmågor (OPTION-begäran). Detta kan vara en fullständig SIP-adress som <b>sip:exempel@exempel.se</b> eller bara användardelen eller telefonnumret av den fullständiga adressen. När du inte anger en fullständig adress så kommer Twinkle att komplettera adressen genom att använda domänvärdet för din användarprofil.Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.&OK&OK&Cancel&AvbrytF10F10TransferFormTwinkle - TransferTwinkle - KopplaTransfer call toKoppla samtal till&To:&Till:The address of the person you want to transfer the call to. This can be a full SIP address like <b>sip:example@example.com</b> or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile.Address bookAdressbokSelect an address from the address book.Välj en adress från adressboken.Type of transferTyp av koppling&Blind transfer&Blind kopplingAlt+BAlt+BTransfer the call to a third party without contacting that third party yourself.T&ransfer with consultationAlt+RBefore transferring the call to a third party, first consult the party yourself.Transfer to other &lineAlt+LAlt+TConnect the remote party on the active line with the remote party on the other line.&OK&OKAlt+OAlt+O&Cancel&AvbrytF10F10TwinkleCoreCannot open file for reading: %1Kan inte öppna fil för läsning: %1File system error while reading file %1 .Fel i filsystemet vid läsning från filen %1.Cannot open file for writing: %1Kan inte öppna fil för skrivning: %1File system error while writing file %1 .Fel i filsystemet vid skrivning till filen %1.AnonymousAnonymWarning:Varning:Failed to create log file %1 .Kunde inte skapa loggfil %1.Excessive number of socket errors.För stort antal uttagsfel (socket).Built with support for:Byggd med stöd för:Contributions:Bidrag:This software contains the following software from 3rd parties:Den här programmet innehåller följande tredjepartsprogramvara:* GSM codec from Jutta Degener and Carsten Bormann, University of Berlin* GSM codec från Jutta Degener och Carsten Bormann vid Berlins universitet* G.711/G.726 codecs from Sun Microsystems (public domain)* G.711/G.726 kodekar från Sun Microsystems (public domain)* iLBC implementation from RFC 3951 (www.ilbcfreeware.org)* iLBC implementation från RFC 3951 (www.ilbcfreeware.org)* Parts of the STUN project at http://sourceforge.net/projects/stun* Delar av STUN-projektet från http://sourceforge.net/projects/stun* Parts of libsrv at http://libsrv.sourceforge.net/* Delar av libsrv från http://libsrv.sourceforge.netFor RTP the following dynamic libraries are linked:För RTP länkas följande dynamiska bibliotek in:Translated to english by <your name>Översatt till svenska av Daniel NylanderDirectory %1 does not exist.Katalogen %1 finns inte.Cannot open file %1 .Kan inte öppna filen %1.%1 is not set to your home directory.%1 är inte satt till din hemkatalog.Directory %1 (%2) does not exist.Katalogen %1 (%2) finns inte.Cannot create directory %1 .Kan inte skapa katalogen %1.Lock file %1 already exist, but cannot be opened.Låsfil %1 finns redan, men kan inte öppnas.%1 is already running.
Lock file %2 already exists.%1 körs redan.
Låsfilen %2 finns redan.Cannot create %1 .Kan inte skapa %1.Cannot write to %1 .Kan inte skriva till %1.Syntax error in file %1 .Syntaxfel i filen %1.Failed to backup %1 to %2Kunde inte kopiera %1 till %2unknown name (device is busy)okänt namn (enheten är upptagen)Default deviceStandardenhetCannot access the ring tone device (%1).Cannot access the speaker (%1).Kan inte komma åt högtalaren (%1).Cannot access the microphone (%1).Kan inte komma åt mikrofonen (%1).Call transfer - %1Sound card cannot be set to full duplex.Ljudkortet kan inte ställas in i full duplex-läge.Cannot set buffer size on sound card.Kan inte ställa in buffertstorlek på ljudkortet.Sound card cannot be set to %1 channels.Ljudkortet kan inte ställas in till %1 kanaler.Cannot set sound card to 16 bits recording.Cannot set sound card to 16 bits playing.Cannot set sound card sample rate to %1Opening ALSA driver failedCannot open ALSA driver for PCM playbackCannot resolve STUN server: %1Kan inte slå upp STUN-server: %1You are behind a symmetric NAT.
STUN will not work.
Configure a public IP address in the user profile
and create the following static bindings (UDP) in your NAT.Du är bakom en symmetrisk NAT.
STUN kommer inte fungera.
Konfigurera en publik IP-adress i användarprofilen
och skapa följande statiska bindningar (UDP) i din NAT.public IP: %1 --> private IP: %2 (SIP signaling)publik IP: %1 --> privat IP: %2 (SIP-signalering)public IP: %1-%2 --> private IP: %3-%4 (RTP/RTCP)publik IP: %1-%2 --> privat IP: %3-%4 (RTP/RTCP)Cannot reach the STUN server: %1Kan inte kontakta STUN-servern: %1If you are behind a firewall then you need to open the following UDP ports.Om du är bakom en brandvägg behöver du öppna följande UDP-portar.Port %1 (SIP signaling)Port %1 (SIP-signalering)Ports %1-%2 (RTP/RTCP)Portar %1-%2 (RTP/RTCP)NAT type discovery via STUN failed.STUN kunde inte detektera NAT-typ.Failed to create file %1Misslyckades med att skapa filen %1Failed to write data to file %1Misslyckades med att skriva data till filen %1Cannot receive incoming TCP connections.Kan inte ta emot inkommande TCP-anslutningar.Cannot open ALSA driver for PCM captureKan inte öppna ALSA-drivrutin för PCM-fångstFailed to send message.Misslyckades med att skicka meddelande.Cannot lock %1 .UserProfileFormTwinkle - User ProfileTwinkle - AnvändarprofilUser profile:Användarprofil:Select which profile you want to edit.Välj den profil du vill ändra.UserAnvändareSIP serverSIP-serverVoice mailRöstbrevlådaRTP audioRTP-ljudSIP protocolSIP-protokollNATNATAddress formatAdressformatTimersRing tonesRingsignalerScriptsSkriptSecuritySäkerhetSelect a category for which you want to see or modify the settings.Välj en kategori för vilken du vill se eller ändra inställningar.&OK&OKAlt+OAlt+OAccept and save your changes.Acceptera och spara dina ändringar.&Cancel&AvbrytAlt+CAlt+AUndo all your changes and close the window.Ångra alla dina ändringar och stäng fönstret.SIP accountSIP-konto&User name*:&Användarnamn*:&Domain*:&Domän*:Or&ganization:Or&ganisation:The SIP user name given to you by your provider. It is the user part in your SIP address, <b>username</b>@domain.com This could be a telephone number.
<br><br>
This field is mandatory.SIP-användarnamnet som din leverantör levererat till dig. Detta är användardelen av din SIP-adress, <b>användarnamn</b>@domän.se. Detta kan vara ett telefonnummer.
<br><br>
Detta fält är obligatoriskt.The domain part of your SIP address, username@<b>domain.com</b>. Instead of a real domain this could also be the hostname or IP address of your <b>SIP proxy</b>. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer.
<br><br>
This field is mandatory.Domändelen av din SIP-adress, användarnamn@<b>domän.se</b>. Istället för en riktig domän så kan detta även vara värdnamnet eller IP-adressen för din <b>SIP-proxy</b>. Om du vill använda direktkommunikation mellan IP-telefon och IP-telefon så kan du fylla i värdnamnet eller IP-adress för din dator.
<br><br>
Detta fält är obligatoriskt.You may fill in the name of your organization. When you make a call, this might be shown to the called party.Du kan fylla i namnet för din organisation. När du ringer ett samtal så kan dock detta visas för motparten.This is just your full name, e.g. John Doe. It is used as a display name. When you make a call, this display name might be shown to the called party.Detta är helt enkelt ditt fullständiga namn, t.ex. Sven Svensson. Det används endast för visning. När du ringer ett samtal kan dock detta namn visas för motparten.&Your name:&Ditt namn:SIP authenticationSIP-autentisering&Realm:Authentication &name:&Password:&Lösenord:The realm for authentication. This value must be provided by your SIP provider. If you leave this field empty, then Twinkle will try the user name and password for any realm that it will be challenged with.Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though.Ditt SIP-autentiseringsnamn. Ganska ofta är detta samma som ditt SIP-användarnamn. Det kan dock vara ett annat namn.Your password for authentication.Ditt lösenord för autentisering.Registrar&Registrar:The hostname, domain name or IP address of your registrar. If you use an outbound proxy that is the same as your registrar, then you may leave this field empty and only fill in the address of the outbound proxy.The registration expiry time that Twinkle will request.secondssekunderRe&gister at startupRe&gistrera vid uppstartAlt+GAlt+GIndicates if Twinkle should automatically register when you run this user profile. You should disable this when you want to do direct IP phone to IP phone communication without a SIP proxy.Outbound ProxyUtgående proxy&Use outbound proxy&Använd utgående proxyAlt+UAlt+AIndicates if Twinkle should use an outbound proxy. If an outbound proxy is used then all SIP requests are sent to this proxy. Without an outbound proxy, Twinkle will try to resolve the SIP address that you type for a call invitation for example to an IP address and send the SIP request there.Outbound &proxy:Utgående &proxy:&Send in-dialog requests to proxyAlt+SAlt+VSIP requests within a SIP dialog are normally sent to the address in the contact-headers exchanged during call setup. If you tick this box, that address is ignored and in-dialog request are also sent to the outbound proxy.&Don't send a request to proxy if its destination can be resolved locally.Alt+DAlt+TWhen you tick this option Twinkle will first try to resolve a SIP address to an IP address itself. If it can, then the SIP request will be sent there. Only when it cannot resolve the address, it will send the SIP request to the proxy (note that an in-dialog request will only be sent to the proxy in this case when you also ticked the previous option.)The hostname, domain name or IP address of your outbound proxy.Värdnamnet, domännamnet eller IP-adressen för din utgående proxy.Co&decsKo&dekarCodecsKodekarAvailable codecs:Tillgängliga kodekar:G.711 A-lawG.711 A-lawG.711 u-lawG.711 u-lawGSMGSMspeex-nb (8 kHz)speex-nb (8 kHz)speex-wb (16 kHz)speex-wb (16 kHz)speex-uwb (32 kHz)speex-uwb (32 kHz)List of available codecs.Lista över tillgängliga kodekar.Move a codec from the list of available codecs to the list of active codecs.Flytta en kodek från listan för tillgängliga kodekar till listan för aktiva kodekar.Move a codec from the list of active codecs to the list of available codecs.Flytta en kodek från listan för aktiva kodekar till listan för tillgängliga kodekar.Active codecs:Aktiva kodekar:List of active codecs. These are the codecs that will be used for media negotiation during call setup. The order of the codecs is the order of preference of use.Move a codec upwards in the list of active codecs, i.e. increase its preference of use.Move a codec downwards in the list of active codecs, i.e. decrease its preference of use.&G.711/G.726 payload size:The preferred payload size for the G.711 and G.726 codecs.msms&Follow codec preference from far end on incoming callsAlt+F<p>
For incoming calls, follow the preference from the far-end (SDP offer). Pick the first codec from the SDP offer that is also in the list of active codecs.
<p>
If you disable this option, then the first codec from the active codecs that is also in the SDP offer is picked.Follow codec &preference from far end on outgoing callsAlt+PAlt+P<p>
For outgoing calls, follow the preference from the far-end (SDP answer). Pick the first codec from the SDP answer that is also in the list of active codecs.
<p>
If you disable this option, then the first codec from the active codecs that is also in the SDP answer is picked.&iLBC&iLBCiLBCiLBCi&LBC payload type:iLBC &payload size (ms):The dynamic type value (96 or higher) to be used for iLBC.20203030The preferred payload size for iLBC.&Speex&SpeexSpeexSpeexPerceptual &enhancementAlt+EAlt+RPerceptual enhancement is a part of the decoder which, when turned on, tries to reduce (the perception of) the noise produced by the coding/decoding process. In most cases, perceptual enhancement make the sound further from the original objectively (if you use SNR), but in the end it still sounds better (subjective improvement).&Ultra wide band payload type:&VAD&VADAlt+VAlt+V&Wide band payload type:V&BRV&BRAlt+BAlt+BVariable bit-rate (VBR) allows a codec to change its bit-rate dynamically to adapt to the "difficulty" of the audio being encoded. In the example of Speex, sounds like vowels and high-energy transients require a higher bit-rate to achieve good quality, while fricatives (e.g. s,f sounds) can be coded adequately with less bits. For this reason, VBR can achieve a lower bit-rate for the same quality, or a better quality for a certain bit-rate. Despite its advantages, VBR has two main drawbacks: first, by only specifying quality, there's no guarantee about the final average bit-rate. Second, for some real-time applications like voice over IP (VoIP), what counts is the maximum bit-rate, which must be low enough for the communication channel.The dynamic type value (96 or higher) to be used for speex wide band.Co&mplexity:DT&XDT&XAlt+XAlt+XDiscontinuous transmission is an addition to VAD/VBR operation, that allows one to stop transmitting completely when the background noise is stationary.The dynamic type value (96 or higher) to be used for speex narrow band.With Speex, it is possible to vary the complexity allowed for the encoder. This is done by controlling how the search is performed with an integer ranging from 1 to 10 in a way that's similar to the -1 to -9 options to gzip and bzip2 compression utilities. For normal use, the noise level at complexity 1 is between 1 and 2 dB higher than at complexity 10, but the CPU requirements for complexity 10 is about 5 times higher than for complexity 1. In practice, the best trade-off is between complexity 2 and 4, though higher settings are often useful when encoding non-speech sounds like DTMF tones.&Narrow band payload type:G.726G.726G.726 &40 kbps payload type:The dynamic type value (96 or higher) to be used for G.726 40 kbps.The dynamic type value (96 or higher) to be used for G.726 32 kbps.G.726 &24 kbps payload type:The dynamic type value (96 or higher) to be used for G.726 24 kbps.G.726 &32 kbps payload type:The dynamic type value (96 or higher) to be used for G.726 16 kbps.G.726 &16 kbps payload type:Codeword &packing order:RFC 3551RFC 3551ATM AAL2ATM AAL2There are 2 standards to pack the G.726 codewords into an RTP packet. RFC 3551 is the default packing method. Some SIP devices use ATM AAL2 however. If you experience bad quality using G.726 with RFC 3551 packing, then try ATM AAL2 packing.DT&MFDT&MFDTMFDTMFThe dynamic type value (96 or higher) to be used for DTMF events (RFC 2833).DTMF vo&lume:The power level of the DTMF tone in dB.The pause after a DTMF tone.DTMF &duration:DTMF payload &type:DTMF &pause:dBdBDuration of a DTMF tone.DTMF t&ransport:DTMF-t&ransport:AutoAutoRFC 2833RFC 2833InbandOut-of-band (SIP INFO)<h2>RFC 2833</h2>
<p>Send DTMF tones as RFC 2833 telephone events.</p>
<h2>Inband</h2>
<p>Send DTMF inband.</p>
<h2>Auto</h2>
<p>If the far end of your call supports RFC 2833, then a DTMF tone will be send as RFC 2833 telephone event, otherwise it will be sent inband.
</p>
<h2>Out-of-band (SIP INFO)</h2>
<p>
Send DTMF out-of-band via a SIP INFO request.
</p>GeneralAllmäntProtocol optionsProtokollalternativRFC 2543RFC 2543RFC 3264RFC 3264Indicates if RFC 2543 (set media IP address in SDP to 0.0.0.0) or RFC 3264 (use direction attributes in SDP) is used to put a call on-hold.Allow m&issing Contact header in 200 OK on REGISTERAlt+IAlt+IA 200 OK response on a REGISTER request must contain a Contact header. Some registrars however, do not include a Contact header or include a wrong Contact header. This option allows for such a deviation from the specs.&Max-Forwards header is mandatoryAlt+MAlt+MAccording to RFC 3261 the Max-Forwards header is mandatory. But many implementations do not send this header. If you tick this box, Twinkle will reject a SIP request if Max-Forwards is missing.Put ®istration expiry time in contact headerAlt+RIn a REGISTER message the expiry time for registration can be put in the Contact header or in the Expires header. If you tick this box it will be put in the Contact header, otherwise it goes in the Expires header.&Use compact header namesIndicates if compact header names should be used for headers that have a compact form.Allow SDP change during call setup<p>A SIP UAS may send SDP in a 1XX response for early media, e.g. ringing tone. When the call is answered the SIP UAS should send the same SDP in the 200 OK response according to RFC 3261. Once SDP has been received, SDP in subsequent responses should be discarded.</p>
<p>By allowing SDP to change during call setup, Twinkle will not discard SDP in subsequent responses and modify the media stream if the SDP is changed. When the SDP in a response is changed, it must have a new version number in the o= line.</p>Use domain &name to create a unique contact header valueAlt+N<p>
Twinkle creates a unique contact header value by combining the SIP user name and domain:
</p>
<p>
<tt> user_domain@local_ip</tt>
</p>
<p>
This way 2 user profiles, having the same user name but different domain names, have unique contact addresses and hence can be activated simultaneously.
</p>
<p>
Some proxies do not handle a contact header value like this. You can disable this option to get a contact header value like this:
</p>
<p>
<tt> user@local_ip</tt>
</p>
<p>
This format is what most SIP phones use.
</p>&Encode Via, Route, Record-Route as listThe Via, Route and Record-Route headers can be encoded as a list of comma separated values or as multiple occurrences of the same header.Redirection&Allow redirectionAlt+AAlt+BIndicates if Twinkle should redirect a request if a 3XX response is received.Ask user &permission to redirectIndicates if Twinkle should ask the user before redirecting a request when a 3XX response is received.The number of redirect addresses that Twinkle tries at a maximum before it gives up redirecting a request. This prevents a request from getting redirected forever.SIP extensionsdisabledinaktiveradsupportedstödsrequiredkrävspreferredföredrasIndicates if the 100rel extension (PRACK) is supported:<br><br>
<b>disabled</b>: 100rel extension is disabled
<br><br>
<b>supported</b>: 100rel is supported (it is added in the supported header of an outgoing INVITE). A far-end can now require a PRACK on a 1xx response.
<br><br>
<b>required</b>: 100rel is required (it is put in the require header of an outgoing INVITE). If an incoming INVITE indicates that it supports 100rel, then Twinkle will require a PRACK when sending a 1xx response. A call will fail when the far-end does not support 100rel.
<br><br>
<b>preferred</b>: Similar to required, but if a call fails because the far-end indicates it does not support 100rel (420 response) then the call will be re-attempted without the 100rel requirement.&100 rel (PRACK):ReplacesErsätterREFERREFERCall transfer (REFER)Alt+TAlt+TIndicates if Twinkle should transfer a call if a REFER request is received.As&k user permission to transferAlt+KIndicates if Twinkle should ask the user before transferring a call when a REFER request is received.Hold call &with referrer while setting up call to transfer targetAlt+WIndicates if Twinkle should put the current call on hold when a REFER request to transfer a call is received.Ho&ld call with referee before sending REFERAlt+LAlt+TIndicates if Twinkle should put the current call on hold when you transfer a call.Auto re&fresh subscription to refer event while call transfer is not finishedWhile a call is being transferred, the referee sends NOTIFY messages to the referrer about the progress of the transfer. These messages are only sent for a short interval which length is determined by the referee. If you tick this box, the referrer will automatically send a SUBSCRIBE to lengthen this interval if it is about to expire and the transfer has not yet been completed.Attended refer to AoR (Address of Record)PrivacyIntegritetPrivacy optionsIntegritetsalternativ&Send P-Preferred-Identity header when hiding user identityInclude a P-Preferred-Identity header with your identity in an INVITE request for a call with identity hiding.NAT traversalNAT-traversering&NAT traversal not neededChoose this option when there is no NAT device between you and your SIP proxy or when your SIP provider offers hosted NAT traversal.&Use statically configured public IP address inside SIP messagesIndicates if Twinkle should use the public IP address specified in the next field inside SIP message, i.e. in SIP headers and SDP body instead of the IP address of your network interface.<br><br>
When you choose this option you have to create static address mappings in your NAT device as well. You have to map the RTP ports on the public IP address to the same ports on the private IP address of your PC.Choose this option when your SIP provider offers a STUN server for NAT traversal.S&TUN server:S&TUN-server:The hostname, domain name or IP address of the STUN server.Värdnamnet, domännamnet eller IP-adressen för STUN-servern.&Public IP address:&Publik IP-adress:The public IP address of your NAT.Telephone numbersTelefonnummerOnly &display user part of URI for telephone numberIf a URI indicates a telephone number, then only display the user part. E.g. if a call comes in from sip:123456@twinklephone.com then display only "123456" to the user. A URI indicates a telephone number if it contains the "user=phone" parameter or when it has a numerical user part and you ticked the next option.&URI with numerical user part is a telephone numberIf you tick this option, then Twinkle considers a SIP address that has a user part that consists of digits, *, #, + and special symbols only as a telephone number. In an outgoing message, Twinkle will add the "user=phone" parameter to such a URI.&Remove special symbols from numerical dial stringsTelephone numbers are often written with special symbols like dashes and brackets to make them readable to humans. When you dial such a number the special symbols must not be dialed. To allow you to simply copy/paste such a number into Twinkle, Twinkle can remove these symbols when you hit the dial button.&Special symbols:The special symbols that may be part of a telephone number for nice formatting, but must be removed when dialing.Number conversionMatch expressionReplaceErsättMove the selected number conversion rule upwards in the list.Move the selected number conversion rule downwards in the list.&Add&Lägg tillAdd a number conversion rule.Re&moveTa &bortRemove the selected number conversion rule.&Edit&RedigeraEdit the selected number conversion rule.Type a telephone number here an press the Test button to see how it is converted by the list of number conversion rules.&Test&TestaTest how a number is converted by the number conversion rules.for STUNför STUNWhen an incoming call is received, this timer is started. If the user answers the call, the timer is stopped. If the timer expires before the user answers the call, then Twinkle will reject the call with a "480 User Not Responding".NAT &keep alive:&No answer:&Inget svar:Ring &back tone:<p>
Specify the file name of a .wav file that you want to be played as ring back tone for this user.
</p>
<p>
This ring back tone overrides the ring back tone settings in the system settings.
</p><p>
Specify the file name of a .wav file that you want to be played as ring tone for this user.
</p>
<p>
This ring tone overrides the ring tone settings in the system settings.
</p>&Ring tone:&Ringsignal:<p>
This script is called when you release a call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing SIP BYE request are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=local_release</b>. <b>SIPREQUEST_METHOD=BYE</b>. <b>SIPREQUEST_URI</b> contains the request-URI of the BYE. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.<p>
This script is called when an incoming call fails.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing SIP failure response are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=in_call_failed</b>. <b>SIPSTATUS_CODE</b> contains the status code of the failure response. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.<p>
This script is called when the remote party releases a call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the incoming SIP BYE request are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=remote_release</b>. <b>SIPREQUEST_METHOD=BYE</b>. <b>SIPREQUEST_URI</b> contains the request-URI of the BYE. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.<p>
You can customize the way Twinkle handles incoming calls. Twinkle can call a script when a call comes in. Based on the output of the script Twinkle accepts, rejects or redirects the call. When accepting the call, the ring tone can be customized by the script as well. The script can be any executable program.
</p>
<p>
<b>Note:</b> Twinkle pauses while your script runs. It is recommended that your script does not take more than 200 ms. When you need more time, you can send the parameters followed by <b>end</b> and keep on running. Twinkle will continue when it receives the <b>end</b> parameter.
</p>
<p>
With your script you can customize call handling by outputting one or more of the following parameters to stdout. Each parameter should be on a separate line.
</p>
<p>
<blockquote>
<tt>
action=[ continue | reject | dnd | redirect | autoanswer ]<br>
reason=<string><br>
contact=<address to redirect to><br>
caller_name=<name of caller to display><br>
ringtone=<file name of .wav file><br>
display_msg=<message to show on display><br>
end<br>
</tt>
</blockquote>
</p>
<h2>Parameters</h2>
<h3>action</h3>
<p>
<b>continue</b> - continue call handling as usual<br>
<b>reject</b> - reject call<br>
<b>dnd</b> - deny call with do not disturb indication<br>
<b>redirect</b> - redirect call to address specified by <b>contact</b><br>
<b>autoanswer</b> - automatically answer a call<br>
</p>
<p>
When the script does not write an action to stdout, then the default action is continue.
</p>
<p>
<b>reason: </b>
With the reason parameter you can set the reason string for reject or dnd. This might be shown to the far-end user.
</p>
<p>
<b>caller_name: </b>
This parameter will override the display name of the caller.
</p>
<p>
<b>ringtone: </b>
The ringtone parameter specifies the .wav file that will be played as ring tone when action is continue.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers in the incoming INVITE message are passed in environment variables to your script. The variable names are formatted as <b>SIP_<HEADER_NAME></b> E.g. SIP_FROM contains the value of the from header.
</p>
<p>
TWINKLE_TRIGGER=in_call. SIPREQUEST_METHOD=INVITE. The request-URI of the INVITE will be passed in <b>SIPREQUEST_URI</b>. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.<p>
This script is called when the remote party answers your call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the incoming 200 OK are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=out_call_answered</b>. <b>SIPSTATUS_CODE=200</b>. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.<p>
This script is called when you answer an incoming call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing 200 OK are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=in_call_answered</b>. <b>SIPSTATUS_CODE=200</b>. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.Call released locall&y:<p>
This script is called when an outgoing call fails.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the incoming SIP failure response are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=out_call_failed</b>. <b>SIPSTATUS_CODE</b> contains the status code of the failure response. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.<p>
This script is called when you make a call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing INVITE are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=out_call</b>. <b>SIPREQUEST_METHOD=INVITE</b>. <b>SIPREQUEST_URI</b> contains the request-URI of the INVITE. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.Outgoing call a&nswered:Incoming call &failed:&Incoming call:&Inkommande samtal:Call released &remotely:Incoming call &answered:O&utgoing call:&Utgående samtal:Out&going call failed:&Enable ZRTP/SRTP encryption&Aktivera ZRTP/SRTP-krypteringZRTP settingsInställningar för ZRTPO&nly encrypt audio if remote party indicated ZRTP support in SDPA SIP endpoint supporting ZRTP may indicate ZRTP support during call setup in its signalling. Enabling this option will cause Twinkle only to encrypt calls when the remote party indicates ZRTP support.&Indicate ZRTP support in SDPTwinkle will indicate ZRTP support during call setup in its signalling.&Popup warning when remote party disables encryption during callA remote party of an encrypted call may send a ZRTP go-clear command to stop encryption. When Twinkle receives this command it will popup a warning if this option is enabled.&Voice mail address:The SIP address or telephone number to access your voice mail.&MWI type:Subscription &duration:Mailbox &user name:The hostname, domain name or IP address of your voice mailbox server.Your user name for accessing your voice mailbox.Mailbox &server:Via outbound &proxyCheck this option if Twinkle should send SIP messages to the mailbox server via the outbound proxy.Dynamic payload type %1 is used more than once.You must fill in a user name for your SIP account.Du måste fylla i ett användarnamn för ditt SIP-konto.You must fill in a domain name for your SIP account.
This could be the hostname or IP address of your PC if you want direct PC to PC dialing.Du måste fylla i ett domännamn för ditt SIP-konto.
Detta kan vara värdnamnet eller IP-adressen för din dator, om du vill ha direktsamtal, PC till PC.Invalid domain.Ogiltig domän.Invalid user name.Ogiltigt användarnamn.Invalid value for registrar.Invalid value for outbound proxy.You must fill in a mailbox user name.You must fill in a mailbox serverInvalid mailbox server.Invalid mailbox user name.Value for public IP address missing.Invalid value for STUN server.Ogiltigt värde för STUN-server.Ring tonesDescription of .wav files in file dialogRingsignalerChoose ring toneVälj ringsignalRing back tonesDescription of .wav files in file dialogAll filesAlla filerChoose incoming call scriptChoose incoming call answered scriptChoose incoming call failed scriptChoose outgoing call scriptChoose outgoing call answered scriptChoose outgoing call failed scriptChoose local release scriptChoose remote release scriptInstant messageSnabbmeddelandePresenceNärvaroTransport/NATTransport/NATAKA AM&FAKA AM&FA&KA OP:A&KA OP:Authentication management field for AKAv1-MD5 authentication.Operator variant key for AKAv1-MD5 authentication.Add q-value to registrationThe q-value indicates the priority of your registered device. If besides Twinkle you register other SIP devices for this account, then the network may use these values to determine which device to try first when delivering a call.The q-value is a value between 0.000 and 1.000. A higher value means a higher priority.SIP transportSIP-transportUDPUDPTCPTCPTransport mode for SIP. In auto mode, the size of a message determines which transport protocol is used. Messages larger than the UDP threshold are sent via TCP. Smaller messages are sent via UDP.T&ransport protocol:T&ransportprotokoll:UDP t&hreshold: bytes byteMessages larger than the threshold are sent via TCP. Smaller messages are sent via UDP.P&ersistent TCP connectionKeep the TCP connection established during registration open such that the SIP proxy can reuse this connection to send incoming requests. Application ping packets are sent to test if the connection is still alive.Use tel-URI for telephone &numberExpand a dialed telephone number to a tel-URI instead of a sip-URI.Select ring back tone file.Select ring tone file.Select script file.&Maximum number of sessions:When you have this number of instant message sessions open, new incoming message sessions will be rejected.&Send composing indications when typing a message.Twinkle sends a composing indication when you type a message. This way the recipient can see that you are typing.Your presenceDin närvaro&Publish availability at startup&Publicera tillgänglighet vid uppstartPublish your availability at startup.Publicera din tillgänglighet vid uppstart.Publication &refresh interval (sec):Refresh rate of presence publications.Buddy presence&Subscription refresh interval (sec):Refresh rate of presence subscriptions.%1 converts to %2%1 konverteras till %2AKA AM&F:Prepr&ocessingPreprocessing (improves quality at remote end)&Automatic gain controlAutomatic gain control (AGC) is a feature that deals with the fact that the recording volume may vary by a large amount between different setups. The AGC provides a way to adjust a signal to a reference volume. This is useful because it removes the need for manual adjustment of the microphone gain. A secondary advantage is that by setting the microphone gain to a conservative (low) level, it is easier to avoid clipping.Automatic gain control &level:Automatic gain control level represents percentual value of automatic gain setting of a microphone. Recommended value is about 25%.&Voice activity detectionWhen enabled, voice activity detection detects whether the input signal represents a speech or a silence/background noise.&Noise reductionThe noise reduction can be used to reduce the amount of background noise present in the input signal. This provides higher quality speech.Acoustic &Echo CancellationIn any VoIP communication, if a speech from the remote end is played in the local loudspeaker, then it propagates in the room and is captured by the microphone. If the audio captured from the microphone is sent directly to the remote end, then the remote user hears an echo of his voice. An acoustic echo cancellation is designed to remove the acoustic echo before it is sent to the remote end. It is important to understand that the echo canceller is meant to improve the quality on the remote end.Variable &bit-rateDiscontinuous &Transmission&Quality:Speex is a lossy codec, which means that it achieves compression at the expense of fidelity of the input speech signal. Unlike some other speech codecs, it is possible to control the tradeoff made between quality and bit-rate. The Speex encoding process is controlled most of the time by a quality parameter that ranges from 0 to 10.Accept call &transfer request (incoming REFER)Allow call transfer while consultation in progressWhen you perform an attended call transfer, you normally transfer the call after you established a consultation call. If you enable this option you can transfer the call while the consultation call is still in progress. This is a non-standard implementation and may not work with all SIP devices.bytesEnable NAT &keep aliveSend UDP NAT keep alive packets.If you have enabled STUN or NAT keep alive, then Twinkle will send keep alive packets at this interval rate to keep the address bindings in your NAT device alive.Do&main*:Organi&zation:E&xpiry:Call Hold &variant:&Max redirections:Indicates if the Replaces-extension is supported.An attended call transfer should use the contact URI as a refer target. A contact URI may not be globally routable however. Alternatively the AoR (Address of Record) may be used. A disadvantage is that the AoR may route to multiple endpoints in case of forking whereas the contact URI routes to a single endpoint.Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.&Send P-Asserted-Identity header when hiding user identityUse STUN (does not wor&k for incoming TCP)STUN ser&ver:<p>
Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings.
</p>
<p>
For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged.
</p>
<p>
The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want.
</p>
<h3>Example 1</h3>
<p>
Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'.
</p>
<p>
The following rules will do the trick:
</p>
<blockquote>
<tt>
Match expression = \+31([0-9]*) , Replace = 0$1<br>
Match expression = \+([0-9]*) , Replace = 00$1</br>
</tt>
</blockquote>
<h3>Example 2</h3>
<p>
You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line.
</p>
<blockquote>
<tt>
Match expression = 0[0-9]* , Replace = 9$&<br>
</tt>
</blockquote>When ZRTP/SRTP is enabled, then Twinkle will try to encrypt the audio of each call you originate or receive. Encryption will only succeed if the remote party has ZRTP/SRTP support enabled. If the remote party does not support ZRTP/SRTP, then the audio channel will stay unencrypted.<H2>Message waiting indication type</H2>
<p>
If your provider offers the message waiting indication service, then Twinkle can show you when new voice mail messages are waiting. Ask your provider which type of message waiting indication is offered.
</p>
<H3>Unsolicited</H3>
<p>
Asterisk provides unsolicited message waiting indication.
</p>
<H3>Solicited</H3>
<p>
Solicited message waiting indication as specified by RFC 3842.
</p>UnsolicitedSolicitedSolicited MWIFor solicited MWI, an endpoint subscribes to the message status for a limited duration. Just before the duration expires, the endpoint should refresh the subscription.WizardFormTwinkle - WizardTwinkle - GuideThe hostname, domain name or IP address of the STUN server.Värdnamnet, domännamnet eller IP-adressen för STUN-servern.S&TUN server:S&TUN-server:The SIP user name given to you by your provider. It is the user part in your SIP address, <b>username</b>@domain.com This could be a telephone number.
<br><br>
This field is mandatory.SIP-användarnamnet som din leverantör levererat till dig. Detta är användardelen av din SIP-adress, <b>användarnamn</b>@domän.se. Detta kan vara ett telefonnummer.
<br><br>
Detta fält är obligatoriskt.&Domain*:&Domän*:Choose your SIP service provider. If your SIP service provider is not in the list, then select <b>Other</b> and fill in the settings you received from your provider.<br><br>
If you select one of the predefined SIP service providers then you only have to fill in your name, user name, authentication name and password.Välj din SIP-tjänsteleverantör. Om din SIP-tjänsteleverantör inte finns med i listan så ska du välja <b>Annan</b> och fylla i inställningarna som du fått från din leverantör.<br><br>
Om du väljer en av de fördefinierade SIP-tjänsteleverantörerna så behöver du endast fylla i ditt namn, användarnamn, autentiseringsnamn och lösenord.&Authentication name:&Autentiseringsnamn:&Your name:&Ditt namn:Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though.Ditt SIP-autentiseringsnamn. Ganska ofta är detta samma som ditt SIP-användarnamn. Det kan dock skilja sig.The domain part of your SIP address, username@<b>domain.com</b>. Instead of a real domain this could also be the hostname or IP address of your <b>SIP proxy</b>. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer.
<br><br>
This field is mandatory.Domändelen av din SIP-adress, användarnamn@<b>domän.se</b>. Istället för en riktig domän så kan detta även vara värdnamnet eller IP-adressen för din <b>SIP-proxy</b>. Om du vill använda direktkommunikation mellan IP-telefon och IP-telefon så kan du fylla i värdnamnet eller IP-adress för din dator.
<br><br>
Detta fält är obligatoriskt.This is just your full name, e.g. John Doe. It is used as a display name. When you make a call, this display name might be shown to the called party.Detta är helt enkelt ditt fullständiga namn, t.ex. Sven Svensson. Det används endast för visning. När du ringer ett samtal kan dock detta namn visas för motparten.SIP pro&xy:SIP-pro&xy:The hostname, domain name or IP address of your SIP proxy. If this is the same value as your domain, you may leave this field empty.Värdnamnet, domännamnet eller IP-adressen för din SIP-proxy. Om detta är samma värde som för din domän så kan du lämna detta fält tomt.&SIP service provider:&SIP-tjänsteleverantör:&Password:&Lösenord:&User name*:A&nvändarnamn*:Your password for authentication.Ditt lösenord för autentisering.&OK&OKAlt+OAlt+O&Cancel&AvbrytAlt+CAlt+ANone (direct IP to IP calls)Ingen (direktsamtal IP till IP)OtherAnnanUser profile wizard:Guide för användarprofil:You must fill in a user name for your SIP account.Du måste fylla i ett användarnamn för ditt SIP-konto.You must fill in a domain name for your SIP account.
This could be the hostname or IP address of your PC if you want direct PC to PC dialing.Du måste fylla i ett domännamn för ditt SIP-konto.
Detta kan vara värdnamnet eller IP-adressen för din dator, om du vill ha direktsamtal, PC till PC.Invalid value for SIP proxy.Ogiltigt värde för SIP-proxy.Invalid value for STUN server.Ogiltigt värde för STUN-server.YesNoDialog&Yes&Ja&No&Nejincoming_callAnswerSvaraRejectAvvisa