AddressCardForm Twinkle - Address Card Twinkle - 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 &OK Alt+O Alt+O &Cancel &Avbryt Alt+C Alt+A You 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. AddressTableModel Name Namn Phone Telefon Remark AuthenticationForm Twinkle - Authentication Twinkle - Autentisering user No need to translate användare The user for which authentication is requested. Användaren för vilken autentisering begärs. profile No need to translate profil The 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 &Avbryt Login required for realm: realm No need to translate The realm for which you need to authenticate. BuddyForm Twinkle - Buddy Twinkle - Kompis Address book Adressbok Select an address from the address book. Välj en adress från adressboken. &Phone: &Telefon: Name of your buddy. Namnet på din kompis. &Show availability &Visa tillgänglighet Alt+S Alt+V Check 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 &OK Alt+O Alt+O &Cancel &Avbryt Alt+C Alt+A You must fill in a name. Du måste ange ett namn. Invalid phone. Ogiltig telefon. Failed to save buddy list: %1 Misslyckades med att spara kompislista: %1 BuddyList Availability Tillgänglighet unknown okänd offline frånkopplad online ansluten request failed begäran misslyckades request rejected begäran avvisades not published inte publicerad failed to publish misslyckades med att publicera Click right to add a buddy. Högerklicka för att lägga till en kompis. CoreAudio Failed to open sound card MIsslyckades med att öppna ljudkortet Failed to create a UDP socket (RTP) on port %1 Misslyckades med att skapa ett UDP-uttag(RTP) på port %1 Failed to create audio receiver thread. Misslyckades med att skapa ljudmottagartråd. Failed to create audio transmitter thread. Misslyckades med att skapa ljudsändartråd. CoreCallHistory local user lokal användare remote user fjärranvändare failure fel unknown okänd in in out ut DeregisterForm Twinkle - Deregister Twinkle - Avregistrering deregister all devices avregistrera alla enheter &OK &OK &Cancel &Avbryt DiamondcardProfileForm 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: &OK &OK &Cancel &Avbryt Fill in your account ID. Fill in your PIN code. A user profile with name %1 already exists. DtmfForm Twinkle - DTMF Twinkle - DTMF Keypad Knappsats 2 2 3 3 Over decadic A. Normally not needed. Över dekadiskt A. Behövs oftast inte. 4 4 5 5 6 6 Over decadic B. Normally not needed. Över dekadiskt B. Behövs oftast inte. 7 7 8 8 9 9 Over decadic C. Normally not needed. Över dekadiskt C. Behövs oftast inte. Star (*) Stjärna (*) 0 0 Pound (#) Fyrkant (#) Over decadic D. Normally not needed. Över dekadiskt D. Behövs oftast inte. 1 1 &Close S&täng Alt+C Alt+T FreeDeskSysTray Show/Hide Visa/Göm Quit Avsluta GUI Failed to create a UDP socket (SIP) on port %1 Kunde inte skapa UDP-socket (SIP) på port %1 Override lock file and start anyway? Strunta i låsfil och starta ändå? The following profiles are both for user %1 Följande profiler är båda för användaren %1 You 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 %2 Linje %1: inkommande samtal för %2 Call transferred by %1 Samtal kopplat av %1 Line %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 %2 Svar på begäran om terminalförmågor: %1 %2 Terminal capabilities of %1 Terminalförmågor för %1 Accepted body types: unknown okänt Accepted encodings: Accepterade kodningar: Accepted languages: Accepterade språk: Allowed requests: Tillåtna begäran: Supported extensions: none End 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 to Linje %1: Skickar vidare till Redirecting request to: %1 Skickar vidare till: %1 Line %1: DTMF detected: Linje %1: DTMF detekterad: invalid DTMF telephone event (%1) Line %1: send DTMF %2 Linje %1: skicka DTMF %2 Line %1: far end does not support DTMF telephone events. Line %1: received notification. Event: %1 Händelse: %1 State: %1 Tillstånd: %1 Reason: %1 Anledning: %1 Progress: %1 %2 Förlopp: %1 %2 Line %1: call transfer failed. Linje %1: samtalskoppling misslyckades. Line %1: call succesfully 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 %2 Linje %1: kopplar samtal till %2 Transfer requested by %1 Koppilng begärd av %1 Line %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 call Vidarekoppling av samtal User 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 request Do you allow the %1 request to be redirected to the following destination? Transferring call Request 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... Abort Avbryt Line %1 Linje %1 Click 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 %2 Misslyckades med att skapa ett %1-uttag (SIP) på port %2 If 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 header Failed to save message attachment: %1 Misslyckades med att spara meddelandebilaga: %1 Accepted by network Accepterad av nätverket Transferred by: %1 Cannot open web browser: %1 Configure your web browser in the system settings. GetAddressForm Twinkle - Select address Twinkle - Välj adress &KAddressBook &KAddressBook Name Namn Type Typ Phone Telefon This list of addresses is taken from KAddressBook. 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-adresser Alt+S Alt+V Check this option when you only want to see contacts with SIP addresses, i.e. starting with "sip:". Kryssa för detta alternativ när du endast vill se kontakter med SIP-adresser, alltså börjar med "sip:". &Reload Upp&datera Alt+R Alt+D Reload the list of addresses from KAddressbook. Uppdatera listan över adresser från KAddressbook. &Local address book &Lokal adressbok Contacts in the local address book of Twinkle. Kontakter i den lokala adressboken för Twinkle. &Add &Lägg till Alt+A Alt+L Add a new contact to the local address book. Lägg till en ny kontakt i den lokala adressboken. &Delete &Ta bort Alt+D Alt+T Delete a contact from the local address book. Ta bort en kontakt från den lokala adressboken. &Edit &Redigera Alt+E Alt+R Edit a contact from the local address book. Redigera en kontakt från den lokala adressboken. &OK &OK Alt+O Alt+O &Cancel &Avbryt Alt+C Alt+A You seem not to have any contacts with a phone number in KAddressBook, KDE's address book application. Twinkle retrieves all contacts with a phone number from KAddressBook. To manage your contacts you have to use KAddressBook.As an alternative you may use Twinkle's local address book. GetProfileNameForm Twinkle - Profile name Twinkle - Profilnamn &OK &OK &Cancel &Avbryt Enter a name for your profile: Ge din profil ett namn: The name of your profile A profile contains your user settings, e.g. your user name and password. You have to give each profile a name. 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. To remember your profiles easily you could use your SIP user name as a profile name, e.g. example@example.com Din profils namn En profil innehåller dina användarinställningar. T.ex. ditt användarnamn och lösenord. Du måste ge varje profil ett namn. 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. För att enkelt komma ihåg dina profiler kan du använda ditt SIP-användarnamn som profilnamn. T.ex. exempel@exempel.com 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: HistoryForm Twinkle - Call History Twinkle - Samtalshistorik Time Tid In/Out In/Ut From/To Från/Till Subject Ämne Status Status Call details Samtalsdetaljer Details of the selected call record. Detaljer för det valda samtalet. View Visa &Incoming calls &Inkommande samtal Alt+I Alt+I Check this option to show incoming calls. Kryssa i denna ruta för att visa inkommande samtal. &Outgoing calls &Utgående samtal Alt+O Alt+U Check this option to show outgoing calls. Kryssa i denna ruta för att visa utgående samtal. &Answered calls &Besvarade samtal Alt+A Alt+B Check this option to show answered calls. Kryssa i denna ruta för att visa besvarade samtal. &Missed calls &Missade samtal Alt+M Alt+M Check this option to show missed calls. Kryssa i denna ruta för att visa missade samtal. Current &user profiles only Enbart aktuell &användarprofil Alt+U Alt+A Check 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 &Töm Alt+L Alt+T Clear the complete call history. Note: this will clear all records, also records not shown depending on the checked view options. Töm hela samtalshistoriken. Observera: detta rensar bort alla samtal. Även samtal som inte visas på grund av aktuella visningsalternativ. &Close &Stäng Alt+C Alt+S Close 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: conversation konversation Call... Ring upp... Delete Ta bort Re: Sv: Clo&se St&äng Alt+S Alt+Ä &Call &Ring Call selected address. Ring markerad adress. Number of calls: ### Total call duration: InviteForm Twinkle - Call Twinkle - 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 book Adressbok Select 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 sip:example@example.com 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 sip:exempel@exempel.com 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 identitet Alt+H Alt+D 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 not hide your IP address. Warning: not all providers support identity hiding. 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 inte din IP-adress. Varning: Inte alla SIP-leverantörer stöder dold identitet. &OK &OK &Cancel &Avbryt Not 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. F10 F10 LogViewForm Twinkle - Log Twinkle - Logg Contents of the current log file (~/.twinkle/twinkle.log) Innehåll i nuvarande loggfil (~/.twinkle/twinkle.log) &Close &Stäng Alt+C Alt+S C&lear &Töm Alt+L Alt+T Clear the log window. This does not clear the log file itself. Töm loggfönstret. Detta tömmer inte själva loggfilen. MessageForm Twinkle - Instant message Twinkle - 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 sip:example@example.com 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 book Adressbok Select an address from the address book. Välj en adress från adressboken. &User profile: &Användarprofil: Conversation Konversation Type your message here and then press "send" to send it. Skriv ditt meddelande här och tryck sedan på "Skicka" för att skicka det. &Send &Skicka Alt+S Alt+S Send the message. Skicka meddelandet. Instant message toolbar Verktygsrad för snabbmeddelanden Send file... Skicka fil... Send file Skicka fil image size is scaled down in preview bildstorleken är nedskalad i förhandsvisning Delivery failure Leverans misslyckades Delivery notification Leveransnotifiering Open 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. F10 F10 Size MessageFormView sending message skickar meddelande MphoneForm Twinkle Twinkle &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 sip:example@example.com 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 sip:exempel@exempel.com 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 book Adressbok Select an address from the address book. Välj en adress från adressboken. Dial Ring Dial 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. Display Line status Linjestatus Line &1: Linje &1: Alt+1 Alt+1 Click to switch to line 1. Klicka för att växla till linje 1. From: Från: To: Till: Subject: Ämne: idle No need to translate Overksam Conference call Konferenssamtal Transferring call sas No need to translate sas Short authentication string g711a/g711a No need to translate g711a/g711a Audio codec Ljudkodek 0:00:00 0:00:00 Call duration Samtalslängd sip:from No need to translate sip:från sip:to No need to translate sip:till subject No need to translate ämne photo No need to translate foto Line &2: Linje &2: Alt+2 Alt+2 Click to switch to line 2. Klicka för att växla till linje 2. &File &Arkiv &Edit &Redigera C&all &Ring Activate line Aktivera linje &Registration &Registrering &Services &Tjänster &View &Visa &Help &Hjälp Quit Avsluta &Quit &Avsluta Ctrl+Q Ctrl+A About Twinkle Om Twinkle &About Twinkle &Om Twinkle Call toolbar text Ring &Call... call menu text &Ring... Call someone Ring någon F5 F5 Answer toolbar text Svara &Answer menu text &Svara Answer incoming call Svara inkommande samtal F6 F6 Bye toolbar text Adjö &Bye menu text &Lägg på Release call Esc Esc Reject toolbar text Avvisa &Reject menu text &Avvisa Reject incoming call Avvisa inkommande samtal F8 F8 &Hold menu text &Parkera Put a call on hold, or retrieve a held call Pakera ett samtal eller återuppta ett parkerat samtal Redirect toolbar text Koppla vidare R&edirect... menu text Koppla vidar&e... Redirect incoming call without answering Koppla vidare inkommande samtal utan att svara Dtmf toolbar text Dtmf &Dtmf... menu text &Dtmf... Open keypad to enter digits for voice menu's Öppna knappsats för att mata in siffror för röstmenyer Register Registrera &Register &Registrera Deregister Avregistrera &Deregister Avre&gistrera Deregister this device Avregistrera denna enhet Show registrations Visa registreringar &Show registrations &Visa registreringar Terminal capabilities Terminalförmågor &Terminal capabilities... menu text &Terminalförmågor... Request terminal capabilities from someone Begär terminalförmågor från någon Do not disturb Stör inte &Do not disturb Stör &inte Call redirection Vidarekoppla samtal Call &redirection... Vidarekoppla sa&mtal... Redial toolbar text Ring igen &Redial menu text Ring &igen Repeat last call Upprepa senaste samtalet F12 F12 About Qt Om Qt About &Qt Om &Qt User profile Användarprofil &User profile... &Användarprofil... Conf toolbar text Konf &Conference menu text &Konferens Join two calls in a 3-way conference Mute toolbar text Tyst &Mute menu text &Tyst Mute a call Stäng av mikrofonen Xfer toolbar text Koppla Trans&fer... menu text Kopp&la... Transfer call Koppla samtal System settings Systeminställningar &System settings... &Systeminställningar... Deregister all Avregistrera alla Deregister &all Avregistrera &alla Deregister all your registered devices Avregistera alla dina registrerade enheter Auto answer Svara automatiskt &Auto answer S&vara automatiskt Log Logg &Log... &Logg... Call history Samtalshistorik Call &history... Samtals&historik... F9 F9 Change user ... Byt användare... &Change user ... &Byt användare... Activate or de-activate users Aktivera eller inaktivera användare What's This? Vad är detta? What's &This? Vad är &detta? Shift+F1 Shift+F1 Line 1 Linje 1 Line 2 Linje 2 idle overksam dialing ringer attempting call, please wait försöker ringa samtal, vänta incoming call inkommande samtal establishing call, please wait etablerar samtal, vänta established etablerat established (waiting for media) etablerat (väntar på media) releasing call, please wait unknown state okänt tillstånd Voice is encrypted Röstkanal är krypterad Click to confirm SAS. Klicka för att bekräfta SAS. Click to clear SAS verification. Transfer consultation User: Användare: Call: Ring: Hide identity Dölj identitet Registration status: Registreringsstatus: Registered Registrerad Failed Misslyckades Not registered Inte registrerad Click 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 meddelanden 1 new message Ett nytt meddelande %1 new messages %1 nya meddelanden 1 old message Ett gammalt meddelande %1 old messages %1 gamla meddelanden Messages waiting Meddelanden väntar No messages Inga meddelanden Voice mail status: Status för röstbrevlåda: Failure Misslyckades Unknown Okänt Click 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/deactivate Klicka för att aktivera/inaktivera Click to activate Klicka för att aktivera Do 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 %1 Följande profiler är båda för användaren %1 You 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 provisioned You 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 list Kompislista You 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 &Display Voice mail Röstbrevlåda &Voice mail &Röstbrevlåda Access voice mail F11 F11 Msg Chatt Instant &message... Snabb&meddelande... Instant message Snabbmeddelande &Buddy list &Kompislista &Call... &Ring... &Edit... &Redigera... &Delete &Ta bort O&ffline Fr&ånkopplad &Online A&nsluten &Change availability &Ändra tillgänglighet &Add buddy... &Lägg till kompis... Failed to save buddy list: %1 Misslyckades med att spara kompislista. %1 F10 F10 Diamondcard Manual &Manual Sign up &Sign up... Recharge... Balance history... Call history... Admin center... Recharge Balance history Admin center Call Ring &Answer &Svara Answer Svara &Bye &Lägg på Bye Adjö &Reject &Avvisa Reject Avvisa &Hold &Parkera Hold R&edirect... Koppla vidar&e... Redirect Koppla vidare &Dtmf... &Dtmf... Dtmf Dtmf &Terminal capabilities... &Terminalförmågor... &Redial Ring &igen Redial Ring igen &Conference &Konferens Conf Konf &Mute &Tyst Mute Tyst Trans&fer... Kopp&la... Xfer Koppla NumberConversionForm Twinkle - Number conversion Twinkle - 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 &OK Alt+O Alt+O &Cancel &Avbryt Alt+C Alt+A Match 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. RedirectForm Twinkle - Redirect Twinkle - Koppla vidare Redirect incoming call to Koppla vidare inkommande samtal till You 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 book Adressbok Select an address from the address book. Välj en adress från adressboken. &OK &OK &Cancel &Avbryt F10 F10 F12 F12 F11 F11 SelectNicForm Twinkle - Select NIC Twinkle - Välj nätverkskort Select 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 &IP Ange som standard-&IP Alt+I Alt+I Make 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 &NIC Ange som standard&nätverkskort Alt+N Alt+N Make 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 &OK Alt+O Alt+O If 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. SelectProfileForm Twinkle - Select user profile Twinkle - Välj användarprofil Select user profile(s) to run: Välj användarprofil(er) att köra: User profile Användarprofil Tick 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 &Ny Alt+N Alt+N Create a new profile with the profile editor. Skapa en ny profil med profilredigeraren. &Wizard &Guide Alt+W Alt+G Create a new profile with the wizard. Skapa en ny profil med guiden. &Edit &Redigera Alt+E Alt+R Edit the highlighted profile. Redigera markerad profil. &Delete &Ta bort Alt+D Alt+T Delete the highlighted profile. Ta bort markerad profil. Ren&ame Byt &namn Alt+A Alt+N Rename the highlighted profile. Byt namn på markerad profil. &Set as default &Ange som standard Alt+S Alt+A Make 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 &Kör Alt+R Alt+K Run Twinkle with the selected profiles. Kör Twinkle med markerade profiler. S&ystem settings S&ysteminställningar Alt+Y Alt+Y Edit the system settings. Redigera systeminställningarna. &Cancel &Avbryt Alt+C Alt+A <html>Before you can use Twinkle, you must create a user profile. Click OK to create a profile.</html> <html>Innan du kan använda Twinkle måste du skapa en användarprofil. 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. 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 profile Ta bort profil Failed to delete profile. Misslyckades med att ta bort profil. Failed to rename profile. Misslyckades med att byta namn på profil. If you want to remove or change the default at a later time, you can do that via the system settings. Cannot find .twinkle directory in your home directory. Kan inte hitta katalogen .twinkle i din hemkatalog. Create profile Ed&itor Alt+I Alt+I Dia&mondcard Alt+M Alt+M Modify profile Startup profile &Diamondcard Create 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. 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. You can create a Diamondcard account to make worldwide calls to regular and cell phones and send SMS messages. Choose what method you wish to use.</html> SelectUserForm Twinkle - Select user Twinkle - Välj användare &Cancel &Avbryt Alt+C Alt+A &Select all &Välj alla Alt+S Alt+V &OK &OK Alt+O Alt+O C&lear all &Töm alla Alt+L Alt+T purpose No need to translate syfte User Användare Register Registrera Select users that you want to register. Välj användare som du vill registrera. Deregister Avregistrera Select users that you want to deregister. Välj användare som du vill avregistrera. Deregister all devices Avregistrera alla enheter Select users for which you want to deregister all devices. Välj användare för vilka du vill avregistrera alla enheter. Do not disturb Stör inte Select users for which you want to enable 'do not disturb'. Välj användare för vilka du vill aktivera 'stör ej'. Auto answer Svara automatiskt Select users for which you want to enable 'auto answer'. Välj användare för vilka du vill aktivera "svara automatiskt". SendFileForm Twinkle - Send File Twinkle - Skicka fil Select file to send. Välj fil att skicka. &File: &Fil: &Subject: &Ämne: &OK &OK Alt+O Alt+O &Cancel &Avbryt Alt+C Alt+A File does not exist. Filen finns inte. Send file... Skicka fil... SrvRedirectForm Twinkle - Call Redirection User: Användare: There are 3 redirect services: Unconditional: redirect all calls Busy: redirect a call if both lines are busy No answer: redirect a call when the no-answer timer expires &Unconditional &Redirect all calls Alt+R Activate the unconditional redirection service. Redirect to You 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 book Adressbok Select an address from the address book. Välj en adress från adressboken. &Busy &Upptagen &Redirect calls when I am busy Activate the redirection when busy service. &No answer &Inget svar &Redirect calls when I do not answer Activate the redirection on no answer service. &OK &OK Alt+O Alt+O Accept and save all changes. Acceptera och spara alla ändringar. &Cancel &Avbryt Alt+C Alt+A Undo your changes and close the window. Ångra dina ändringar och stäng fönstret. You have entered an invalid destination. Du har angivit en ogiltig destination. F10 F10 F11 F11 F12 F12 SysSettingsForm Twinkle - System Settings Twinkle - Systeminställningar General Generellt Audio Ljud Ring tones Ringsignaler Address book Adressbok Network Nätverk Log Logg Select a category for which you want to see or modify the settings. Välj den kategori som du vill titta eller ändra inställningar för. &OK &OK Alt+O Alt+O Accept and save your changes. Acceptera och spara dina ändringar. &Cancel &Avbryt Alt+C Alt+A Undo all your changes and close the window. Ångra alla ändringar och stäng fönstret. Sound Card Ljudkort Select 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 användning Alt+V Alt+V Twinkle validates the audio devices before usage to avoid an established call without an audio channel. On startup of Twinkle a warning is given if an audio device is inaccessible. If before making a call, the microphone or speaker appears to be invalid, a warning is given and no call can be made. 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 microphone Reducera &brus från mikrofonen Alt+N Alt+B Advanced Avancerat OSS &fragment size: 16 16 32 32 64 64 128 128 256 256 The 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. MB MB Log &debug reports Alt+D Alt+T Indicates if reports marked as "debug" will be logged. Log &SIP reports Logga &SIP-rapporter Alt+S Alt+S Indicates if SIP messages will be logged. Log S&TUN reports Logga S&TUN-rapporter Alt+T Alt+T Indicates if STUN messages will be logged. Log m&emory reports Alt+E Alt+R Indicates if reports concerning memory management will be logged. System tray Aktivitetsfält Create &system tray icon on startup Skapa ikon i a&ktivitetsfält vid uppstart Enable 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 window Alt+H Alt+D Enable this option if you want Twinkle to hide in the system tray when you close the main window. Startup Uppstart S&tartup hidden in system tray Next 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 profiles Användarprofiler som standard If 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. Services Tjänster Call &waiting Samtal &väntar Alt+W Alt+V With 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+B Alt+B Hang 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 after Alt+A Alt+B When 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. secs s &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 tone Ringsignal &Play ring tone on incoming call S&pela upp ringsignal vid inkommande samtal Alt+P Alt+P Indicates 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 tone Stan&dardringsignal Play the default ring tone when a call comes in. Spela upp standardringsignalen när ett samtal kommer in. C&ustom ring tone A&npassad ringsignal Alt+U Alt+N Play 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 tone P&lay ring back tone when network does not play ring back tone Alt+L Alt+T Play ring back tone while you are waiting for the far-end to answer your call. Depending on your SIP provider the network might provide ring back tone or an announcement. D&efault ring back tone Play the default ring back tone. Cu&stom ring back tone Play 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 call On 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 name Alt+R The 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 call Slå upp &foto för inkommande samtal Lookup the photo of a caller in your address book and display it on an incoming call. none This is the 'none' in default IP address combo ingen none This is the 'none' in default network interface combo ingen Ring tones Description of .wav files in file dialog Ringsignaler Choose ring tone Välj ringsignal Ring back tones Description of .wav files in file dialog Choose ring back tone Maximum 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. 512 512 1024 1024 Tip: for crackling sound with PulseAudio, set play period size to maximum. SysTrayPopup Answer Svara Reject Avvisa Incoming Call TermCapForm Twinkle - Terminal Capabilities Twinkle - Terminalförmågor &From: &Från: Get terminal capabilities of Hämta terminalförmågor för &To: &Till: The address that you want to query for capabilities (OPTION request). This can be a full SIP address like sip:example@example.com 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 sip:exempel@exempel.se 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 book Adressbok Select an address from the address book. Välj en adress från adressboken. &OK &OK &Cancel &Avbryt F10 F10 TransferForm Twinkle - Transfer Twinkle - Koppla Transfer call to Koppla samtal till &To: &Till: The address of the person you want to transfer the call to. This can be a full SIP address like sip:example@example.com 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 book Adressbok Select an address from the address book. Välj en adress från adressboken. Type of transfer Typ av koppling &Blind transfer &Blind koppling Alt+B Alt+B Transfer the call to a third party without contacting that third party yourself. T&ransfer with consultation Alt+R Before transferring the call to a third party, first consult the party yourself. Transfer to other &line Alt+L Alt+T Connect the remote party on the active line with the remote party on the other line. &OK &OK Alt+O Alt+O &Cancel &Avbryt F10 F10 TwinkleCore Cannot open file for reading: %1 Kan inte öppna fil för läsning: %1 File system error while reading file %1 . Fel i filsystemet vid läsning från filen %1. Cannot open file for writing: %1 Kan inte öppna fil för skrivning: %1 File system error while writing file %1 . Fel i filsystemet vid skrivning till filen %1. Anonymous Anonym Warning: 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.net For 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 Nylander Directory %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 %2 Kunde inte kopiera %1 till %2 unknown name (device is busy) okänt namn (enheten är upptagen) Default device Standardenhet Cannot 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 - %1 Sound 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 %1 Opening ALSA driver failed Cannot open ALSA driver for PCM playback Cannot resolve STUN server: %1 Kan inte slå upp STUN-server: %1 You 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: %1 Kan inte kontakta STUN-servern: %1 If 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 %1 Misslyckades med att skapa filen %1 Failed to write data to file %1 Misslyckades med att skriva data till filen %1 Cannot receive incoming TCP connections. Kan inte ta emot inkommande TCP-anslutningar. Cannot open ALSA driver for PCM capture Kan inte öppna ALSA-drivrutin för PCM-fångst Failed to send message. Misslyckades med att skicka meddelande. Cannot lock %1 . UserProfileForm Twinkle - User Profile Twinkle - Användarprofil User profile: Användarprofil: Select which profile you want to edit. Välj den profil du vill ändra. User Användare SIP server SIP-server Voice mail Röstbrevlåda RTP audio RTP-ljud SIP protocol SIP-protokoll NAT NAT Address format Adressformat Timers Ring tones Ringsignaler Scripts Skript Security Säkerhet Select a category for which you want to see or modify the settings. Välj en kategori för vilken du vill se eller ändra inställningar. &OK &OK Alt+O Alt+O Accept and save your changes. Acceptera och spara dina ändringar. &Cancel &Avbryt Alt+C Alt+A Undo all your changes and close the window. Ångra alla dina ändringar och stäng fönstret. SIP account SIP-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, username@domain.com This could be a telephone number. This field is mandatory. SIP-användarnamnet som din leverantör levererat till dig. Detta är användardelen av din SIP-adress, användarnamn@domän.se. Detta kan vara ett telefonnummer. Detta fält är obligatoriskt. The domain part of your SIP address, username@domain.com. Instead of a real domain this could also be the hostname or IP address of your SIP proxy. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer. This field is mandatory. Domändelen av din SIP-adress, användarnamn@domän.se. Istället för en riktig domän så kan detta även vara värdnamnet eller IP-adressen för din SIP-proxy. Om du vill använda direktkommunikation mellan IP-telefon och IP-telefon så kan du fylla i värdnamnet eller IP-adress för din dator. 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 authentication SIP-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. &Expiry: The registration expiry time that Twinkle will request. seconds sekunder Re&gister at startup Re&gistrera vid uppstart Alt+G Alt+G Indicates 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 Proxy Utgående proxy &Use outbound proxy &Använd utgående proxy Alt+U Alt+A Indicates 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 proxy Alt+S Alt+V SIP 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+D Alt+T When 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&decs Ko&dekar Codecs Kodekar Available codecs: Tillgängliga kodekar: G.711 A-law G.711 A-law G.711 u-law G.711 u-law GSM GSM speex-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. ms ms &Follow codec preference from far end on incoming calls Alt+F 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. 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 calls Alt+P Alt+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. If you disable this option, then the first codec from the active codecs that is also in the SDP answer is picked. &iLBC &iLBC iLBC iLBC i&LBC payload type: iLBC &payload size (ms): The dynamic type value (96 or higher) to be used for iLBC. 20 20 30 30 The preferred payload size for iLBC. &Speex &Speex Speex Speex Perceptual &enhancement Alt+E Alt+R Perceptual 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 &VAD Alt+V Alt+V &Wide band payload type: V&BR V&BR Alt+B Alt+B Variable 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&X DT&X Alt+X Alt+X Discontinuous transmission is an addition to VAD/VBR operation, that allows 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.726 G.726 G.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 3551 RFC 3551 ATM AAL2 ATM AAL2 There 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&MF DT&MF DTMF DTMF The 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: dB dB Duration of a DTMF tone. DTMF t&ransport: DTMF-t&ransport: Auto Auto RFC 2833 RFC 2833 Inband Out-of-band (SIP INFO) <h2>RFC 2833</h2> Send DTMF tones as RFC 2833 telephone events. <h2>Inband</h2> Send DTMF inband. <h2>Auto</h2> 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. <h2>Out-of-band (SIP INFO)</h2> Send DTMF out-of-band via a SIP INFO request. General Allmänt Protocol options Protokollalternativ Call &Hold variant: RFC 2543 RFC 2543 RFC 3264 RFC 3264 Indicates 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 REGISTER Alt+I Alt+I A 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 mandatory Alt+M Alt+M According 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 &registration expiry time in contact header Alt+R In 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 names Indicates if compact header names should be used for headers that have a compact form. Allow SDP change during call setup 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. 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. Use domain &name to create a unique contact header value Alt+N Twinkle creates a unique contact header value by combining the SIP user name and domain: <tt> user_domain@local_ip</tt> This way 2 user profiles, having the same user name but different domain names, have unique contact addresses and hence can be activated simultaneously. Some proxies do not handle a contact header value like this. You can disable this option to get a contact header value like this: <tt> user@local_ip</tt> This format is what most SIP phones use. &Encode Via, Route, Record-Route as list The 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 redirection Alt+A Alt+B Indicates if Twinkle should redirect a request if a 3XX response is received. Ask user &permission to redirect Indicates if Twinkle should ask the user before redirecting a request when a 3XX response is received. Max re&directions: 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 extensions disabled inaktiverad supported stöds required krävs preferred föredras Indicates if the 100rel extension (PRACK) is supported: disabled: 100rel extension is disabled supported: 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. required: 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. preferred: 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): Replaces Ersätter Indicates if the Replaces-extenstion is supported. REFER REFER Call transfer (REFER) Alt+T Alt+T Indicates if Twinkle should transfer a call if a REFER request is received. As&k user permission to transfer Alt+K Indicates 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 target Alt+W Indicates 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 REFER Alt+L Alt+T Indicates 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 finished While 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) 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 endoint. Privacy Integritet Privacy options Integritetsalternativ &Send P-Preferred-Identity header when hiding user identity Include a P-Preferred-Identity header with your identity in an INVITE request for a call with identity hiding. NAT traversal NAT-traversering &NAT traversal not needed Choose 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 messages Indicates 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. 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 numbers Telefonnummer Only &display user part of URI for telephone number If 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 number If 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 strings Telephone 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 conversion Match expression Replace Ersätt Often the format of the telphone 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. 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. The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. <h3>Example 1</h3> 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'. The following rules will do the trick: <blockquote> <tt> Match expression = \+31([0-9]*) , Replace = 0$1 Match expression = \+([0-9]*) , Replace = 00$1 </tt> </blockquote> <h3>Example 2</h3> You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. <blockquote> <tt> Match expression = 0[0-9]* , Replace = 9$& </tt> </blockquote> Move the selected number conversion rule upwards in the list. Move the selected number conversion rule downwards in the list. &Add &Lägg till Add a number conversion rule. Re&move Ta &bort Remove the selected number conversion rule. &Edit &Redigera Edit 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 &Testa Test how a number is converted by the number conversion rules. for STUN för STUN When 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: Specify the file name of a .wav file that you want to be played as ring back tone for this user. This ring back tone overrides the ring back tone settings in the system settings. Specify the file name of a .wav file that you want to be played as ring tone for this user. This ring tone overrides the ring tone settings in the system settings. &Ring tone: &Ringsignal: This script is called when you release a call. <h2>Environment variables</h2> The values of all SIP headers of the outgoing SIP BYE request are passed in environment variables to your script. TWINKLE_TRIGGER=local_release. SIPREQUEST_METHOD=BYE. SIPREQUEST_URI contains the request-URI of the BYE. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when an incoming call fails. <h2>Environment variables</h2> The values of all SIP headers of the outgoing SIP failure response are passed in environment variables to your script. TWINKLE_TRIGGER=in_call_failed. SIPSTATUS_CODE contains the status code of the failure response. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when the remote party releases a call. <h2>Environment variables</h2> The values of all SIP headers of the incoming SIP BYE request are passed in environment variables to your script. TWINKLE_TRIGGER=remote_release. SIPREQUEST_METHOD=BYE. SIPREQUEST_URI contains the request-URI of the BYE. The name of the user profile will be passed in TWINKLE_USER_PROFILE. You can customize the way Twinkle handles incoming calls. Twinkle can call a script when a call comes in. Based on the ouput 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. Note: 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 end and keep on running. Twinkle will continue when it receives the end parameter. With your script you can customize call handling by outputing one or more of the following parameters to stdout. Each parameter should be on a separate line. <blockquote> <tt> action=[ continue | reject | dnd | redirect | autoanswer ] reason=<string> contact=<address to redirect to> caller_name=<name of caller to display> ringtone=<file name of .wav file> display_msg=<message to show on display> end </tt> </blockquote> <h2>Parameters</h2> <h3>action</h3> continue - continue call handling as usual reject - reject call dnd - deny call with do not disturb indication redirect - redirect call to address specified by contact autoanswer - automatically answer a call When the script does not write an action to stdout, then the default action is continue. reason: With the reason parameter you can set the reason string for reject or dnd. This might be shown to the far-end user. caller_name: This parameter will override the display name of the caller. ringtone: The ringtone parameter specifies the .wav file that will be played as ring tone when action is continue. <h2>Environment variables</h2> 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 SIP_<HEADER_NAME> E.g. SIP_FROM contains the value of the from header. TWINKLE_TRIGGER=in_call. SIPREQUEST_METHOD=INVITE. The request-URI of the INVITE will be passed in SIPREQUEST_URI. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when the remote party answers your call. <h2>Environment variables</h2> The values of all SIP headers of the incoming 200 OK are passed in environment variables to your script. TWINKLE_TRIGGER=out_call_answered. SIPSTATUS_CODE=200. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when you answer an incoming call. <h2>Environment variables</h2> The values of all SIP headers of the outgoing 200 OK are passed in environment variables to your script. TWINKLE_TRIGGER=in_call_answered. SIPSTATUS_CODE=200. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. Call released locall&y: This script is called when an outgoing call fails. <h2>Environment variables</h2> The values of all SIP headers of the incoming SIP failure response are passed in environment variables to your script. TWINKLE_TRIGGER=out_call_failed. SIPSTATUS_CODE contains the status code of the failure response. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when you make a call. <h2>Environment variables</h2> The values of all SIP headers of the outgoing INVITE are passed in environment variables to your script. TWINKLE_TRIGGER=out_call. SIPREQUEST_METHOD=INVITE. SIPREQUEST_URI contains the request-URI of the INVITE. The name of the user profile will be passed in TWINKLE_USER_PROFILE. 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-kryptering 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 unecrypted. ZRTP settings Inställningar för ZRTP O&nly encrypt audio if remote party indicated ZRTP support in SDP A 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 SDP Twinkle will indicate ZRTP support during call setup in its signalling. &Popup warning when remote party disables encryption during call A 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. Unsollicited Sollicited <H2>Message waiting indication type</H2> 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. <H3>Unsollicited</H3> Asterisk provides unsollicited message waiting indication. <H3>Sollicited</H3> Sollicited message waiting indication as specified by RFC 3842. &MWI type: Sollicited MWI Subscription &duration: Mailbox &user name: The hostname, domain name or IP address of your voice mailbox server. For sollicited MWI, an endpoint subscribes to the message status for a limited duration. Just before the duration expires, the endpoint should refresh the subscription. Your user name for accessing your voice mailbox. Mailbox &server: Via outbound &proxy Check 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 server Invalid mailbox server. Invalid mailbox user name. Value for public IP address missing. Invalid value for STUN server. Ogiltigt värde för STUN-server. Ring tones Description of .wav files in file dialog Ringsignaler Choose ring tone Välj ringsignal Ring back tones Description of .wav files in file dialog All files Alla filer Choose incoming call script Choose incoming call answered script Choose incoming call failed script Choose outgoing call script Choose outgoing call answered script Choose outgoing call failed script Choose local release script Choose remote release script Instant message Snabbmeddelande Presence Närvaro Transport/NAT Transport/NAT AKA AM&F AKA AM&F A&KA OP: A&KA OP: Authentication management field for AKAv1-MD5 authentication. Operator variant key for AKAv1-MD5 authentication. Add q-value to registration The 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 transport SIP-transport UDP UDP TCP TCP Transport 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 byte Messages larger than the threshold are sent via TCP. Smaller messages are sent via UDP. Use &STUN (does not work for incoming TCP) P&ersistent TCP connection Keep 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 &number Expand 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 presence Din närvaro &Publish availability at startup &Publicera tillgänglighet vid uppstart Publish 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 %2 AKA AM&F: Prepr&ocessing Preprocessing (improves quality at remote end) &Automatic gain control Automatic 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 detection When enabled, voice activity detection detects whether the input signal represents a speech or a silence/background noise. &Noise reduction The noise reduction can be used to reduce the amount of background noise present in the input signal. This provides higher quality speech. Acoustic &Echo Cancellation In 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-rate Discontinuous &Transmission &Quality: Speex is a lossy codec, which means that it achives 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 progress When 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. bytes Enable NAT &keep alive Send 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. WizardForm Twinkle - Wizard Twinkle - Guide The 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, username@domain.com This could be a telephone number. This field is mandatory. SIP-användarnamnet som din leverantör levererat till dig. Detta är användardelen av din SIP-adress, användarnamn@domän.se. Detta kan vara ett telefonnummer. Detta fält är obligatoriskt. &Domain*: &Domän*: Choose your SIP service provider. If your SIP service provider is not in the list, then select Other and fill in the settings you received from your provider. 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 Annan och fylla i inställningarna som du fått från din leverantör. 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@domain.com. Instead of a real domain this could also be the hostname or IP address of your SIP proxy. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer. This field is mandatory. Domändelen av din SIP-adress, användarnamn@domän.se. Istället för en riktig domän så kan detta även vara värdnamnet eller IP-adressen för din SIP-proxy. Om du vill använda direktkommunikation mellan IP-telefon och IP-telefon så kan du fylla i värdnamnet eller IP-adress för din dator. 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 &OK Alt+O Alt+O &Cancel &Avbryt Alt+C Alt+A None (direct IP to IP calls) Ingen (direktsamtal IP till IP) Other Annan User 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 &Nej