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Diffstat (limited to 'src/gui/lang/twinkle_xx.ts')
| -rw-r--r-- | src/gui/lang/twinkle_xx.ts | 5669 |
1 files changed, 5669 insertions, 0 deletions
diff --git a/src/gui/lang/twinkle_xx.ts b/src/gui/lang/twinkle_xx.ts new file mode 100644 index 0000000..6f2ade3 --- /dev/null +++ b/src/gui/lang/twinkle_xx.ts @@ -0,0 +1,5669 @@ +<!DOCTYPE TS><TS> +<context> + <name>AddressCardForm</name> + <message> + <source>Twinkle - Address Card</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Remark:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Infix name of contact.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>First name of contact.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&First name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You may place any remark about the contact here.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Phone:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Infix name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Phone number or SIP address of contact.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Last name of contact.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Last name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a name.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a phone number or SIP address.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>AuthenticationForm</name> + <message> + <source>Twinkle - Authentication</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>user</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>The user for which authentication is requested.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>profile</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>The user profile of the user for which authentication is requested.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User profile:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Password:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your password for authentication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&User name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Login required for realm:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>realm</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>The realm for which you need to authenticate.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>BuddyForm</name> + <message> + <source>Twinkle - Buddy</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select an address from the address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Phone:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Name of your buddy.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Show availability</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+S</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option if you want to see the availability of your buddy. This will only work if your provider offers a presence agent.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP address your buddy.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a name.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid phone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to save buddy list: %1</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>BuddyList</name> + <message> + <source>Availability</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>unknown</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>offline</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>online</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>request failed</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>request rejected</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>not published</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>failed to publish</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Click right to add a buddy.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>CoreAudio</name> + <message> + <source>Failed to open sound card</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to create a UDP socket (RTP) on port %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to create audio receiver thread.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to create audio transmitter thread.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>CoreCallHistory</name> + <message> + <source>local user</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>remote user</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>failure</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>unknown</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>in</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>out</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>DeregisterForm</name> + <message> + <source>Twinkle - Deregister</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>deregister all devices</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>DiamondcardProfileForm</name> + <message> + <source>Twinkle - Diamondcard User Profile</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your Diamondcard account ID.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Account ID:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&PIN code:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Your name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p align="center"><u>Sign up for a Diamondcard account</u></p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Fill in your account ID.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Fill in your PIN code.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>A user profile with name %1 already exists.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your Diamondcard PIN code.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p>With a Diamondcard account you can make worldwide calls to regular and cell phones and send SMS messages. To sign up for a Diamondcard account click on the "sign up" link below. Once you have signed up you receive an account ID and PIN code. Enter the account ID and PIN code below to create a Twinkle user profile for your Diamondcard account.</p> +<p>For call rates see the sign up web page that will be shown to you when you click on the "sign up" link.</p></source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>DtmfForm</name> + <message> + <source>Twinkle - DTMF</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Keypad</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>3</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Over decadic A. Normally not needed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>4</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>5</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>6</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Over decadic B. Normally not needed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>7</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>8</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>9</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Over decadic C. Normally not needed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Star (*)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>0</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Pound (#)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Over decadic D. Normally not needed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Close</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>FreeDeskSysTray</name> + <message> + <source>Show/Hide</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Quit</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>GUI</name> + <message> + <source>Failed to create a %1 socket (SIP) on port %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Override lock file and start anyway?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The following profiles are both for user %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You can only run multiple profiles for different users.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>If these are users for different domains, then enable the following option in your user profile (SIP protocol)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use domain name to create a unique contact header</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: incoming call for %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call transferred by %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: far end cancelled call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: far end released call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: SDP answer from far end not supported.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: SDP answer from far end missing.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: Unsupported content type in answer from far end.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: no ACK received, call will be terminated.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: no PRACK received, call will be terminated.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: PRACK failed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: failed to cancel call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: far end answered call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call failed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The call can be redirected to:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call released.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call established.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Response on terminal capability request: %1 %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Terminal capabilities of %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accepted body types:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>unknown</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accepted encodings:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accepted languages:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Allowed requests:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Supported extensions:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>none</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>End point type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call retrieve failed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, registration failed: %2 %3</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, registration succeeded (expires = %2 seconds)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, registration failed: STUN failure</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, de-registration succeeded: %2 %3</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, de-registration failed: %2 %3</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, fetching registrations failed: %2 %3</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>: you are not registered</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>: you have the following registrations</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>: fetching registrations...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: redirecting request to</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redirecting request to: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: DTMF detected:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>invalid DTMF telephone event (%1)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: send DTMF %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: far end does not support DTMF telephone events.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: received notification.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Event: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>State: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Reason: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Progress: %1 %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call transfer failed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call succesfully transferred.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call transfer still in progress.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>No further notifications will be received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: transferring call to %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transfer requested by %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: Call transfer failed. Retrieving original call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, STUN request failed: %2 %3</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, STUN request failed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redirecting call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User profile:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Do you allow the call to be redirected to the following destination?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redirecting request</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Do you allow the %1 request to be redirected to the following destination?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transferring call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Request to transfer call received from:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Request to transfer call received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Do you allow the call to be transferred to the following destination?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Info:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Warning:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Critical:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Firewall / NAT discovery...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Abort</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Click the padlock to confirm a correct SAS.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The remote user on line %1 disabled the encryption.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: SAS confirmed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: SAS confirmation reset.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, voice mail status failure.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, voice mail status rejected.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, voice mailbox does not exist.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1, voice mail status terminated.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accepted by network</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call rejected.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line %1: call redirected.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to start conference.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to save message attachment: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transferred by: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot open web browser: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Configure your web browser in the system settings.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>GetAddressForm</name> + <message> + <source>Twinkle - Select address</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&KAddressBook</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Name</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Type</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Phone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>This 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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Show only SIP addresses</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+S</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option when you only want to see contacts with SIP addresses, i.e. starting with "<b>sip:</b>".</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Reload</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+R</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Reload the list of addresses from KAddressbook.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Local address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Remark</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Contacts in the local address book of Twinkle.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Add</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+A</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Add a new contact to the local address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Delete</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+D</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete a contact from the local address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Edit</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+E</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Edit a contact from the local address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>GetProfileNameForm</name> + <message> + <source>Twinkle - Profile name</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Enter a name for your profile:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><b>The name of your profile</b> +<br><br> +A profile contains your user settings, e.g. your user name and password. You have to give each profile a name. +<br><br> +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. +<br><br> +To remember your profiles easily you could use your SIP user name as a profile name, e.g. <b>example@example.com</b></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot find .twinkle directory in your home directory.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Profile already exists.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Rename profile '%1' to:</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>HistoryForm</name> + <message> + <source>Twinkle - Call History</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Time</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>In/Out</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>From/To</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Subject</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Status</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call details</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Details of the selected call record.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>View</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Incoming calls</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+I</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option to show incoming calls.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Outgoing calls</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option to show outgoing calls.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Answered calls</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+A</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option to show answered calls.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Missed calls</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+M</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option to show missed calls.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Current &user profiles only</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+U</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option to show only calls associated with this user profile.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>C&lear</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+L</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Clo&se</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+S</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Close this window.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call selected address.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call start:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call answer:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call end:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call duration:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Direction:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>From:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>To:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Reply to:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Referred by:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Subject:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Released by:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Status:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Far end device:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User profile:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>conversation</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Re:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Number of calls:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>###</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Total call duration:</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>InviteForm</name> + <message> + <source>Twinkle - Call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&To:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Optionally you can provide a subject here. This might be shown to the callee.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select an address from the address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The user that will make the call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Subject:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&From:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Hide identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+H</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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. +</p> +<p> +<b>Warning:</b> not all providers support identity hiding. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Not all SIP providers support identity hiding. Make sure your SIP provider supports it if you really need it.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F10</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>LogViewForm</name> + <message> + <source>Twinkle - Log</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Contents of the current log file (~/.twinkle/twinkle.log)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Close</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>C&lear</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+L</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Clear the log window. This does <b>not</b> clear the log file itself.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>MessageForm</name> + <message> + <source>Twinkle - Instant message</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&To:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The user that will send the message.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select an address from the address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&User profile:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Conversation</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Type your message here and then press "send" to send it.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+S</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Send the message.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delivery failure</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delivery notification</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Instant message toolbar</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Send file...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Send file</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>image size is scaled down in preview</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Open with %1...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Open with...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Save attachment as...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>File already exists. Do you want to overwrite this file?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to save attachment.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1 is typing a message.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F10</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Size</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>MessageFormView</name> + <message> + <source>sending message</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>MphoneForm</name> + <message> + <source>Twinkle</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Buddy list</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Call:</source> + <comment>Label in front of combobox to enter address</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select an address from the address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Dial</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Dial the address.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&User:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The user that will make the call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Auto answer indication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call redirect indication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Do not disturb indication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Message waiting indication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Missed call indication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Registration status.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Display</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line status</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line &1:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Click to switch to line 1.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>From:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>To:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Subject:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Visual indication of line state.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>idle</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call is on hold</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Voice is muted</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Conference call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transferring call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +The padlock indicates that your voice is encrypted during transport over the network. +</p> +<h3>SAS - Short Authentication String</h3> +<p> +Both ends of an encrypted voice channel receive the same SAS on the first call. If the SAS is different at each end, your voice channel may be compromised by a man-in-the-middle attack (MitM). +</p> +<p> +If the SAS is equal at both ends, then you should confirm it by clicking this padlock for stronger security of future calls to the same destination. For subsequent calls to the same destination, you don't have to confirm the SAS again. The padlock will show a check symbol when the SAS has been confirmed. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>sas</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Short authentication string</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>g711a/g711a</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Audio codec</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>0:00:00</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call duration</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>sip:from</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>sip:to</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>subject</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>photo</source> + <comment>No need to translate</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Line &2:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Click to switch to line 2.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&File</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Edit</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>C&all</source> + <translation 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<translation type="unfinished"></translation> + </message> + <message> + <source>About Twinkle</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&About Twinkle</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Call...</source> + <comment>call menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call someone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F5</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Answer</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Answer</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Answer incoming call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F6</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Bye</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Bye</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Release call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Esc</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Reject</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Reject</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Reject incoming call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F8</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Hold</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Hold</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Put a call on hold, or retrieve a held call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redirect</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>R&edirect...</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redirect incoming call without answering</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Dtmf</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Dtmf...</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Open keypad to enter digits for voice menu's</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Register</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Register</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Deregister</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Deregister</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Deregister this device</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Show registrations</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Show registrations</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Terminal capabilities</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Terminal capabilities...</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Request terminal capabilities from someone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Do not disturb</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Do not disturb</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call redirection</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call &redirection...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redial</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Redial</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Repeat last call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F12</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>About Qt</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>About &Qt</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User profile</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&User profile...</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Conf</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Conference</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Join two calls in a 3-way conference</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Mute</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Mute</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Mute a call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Xfer</source> + <comment>toolbar text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Trans&fer...</source> + <comment>menu text</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transfer call</source> + <translation type="unfinished"></translation> + </message> + <message> + 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System Settings</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>General</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Audio</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring tones</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Network</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Log</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select a category for which you want to see or modify the settings.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accept and save your changes.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Undo all your changes and close the window.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Sound Card</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select the sound card for playing the ring tone for incoming calls.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select the sound card to which your microphone is connected.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select the sound card for the speaker function during a call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Speaker:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Ring tone:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Other device:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Microphone:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Validate devices before usage</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+V</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Advanced</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>OSS &fragment size:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>16</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>32</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>64</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>128</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>256</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>ALSA &play period size:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&ALSA capture period size:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max log size:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>MB</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Log &debug reports</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+D</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if reports marked as "debug" will be logged.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Log &SIP reports</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+S</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if SIP messages will be logged.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Log S&TUN reports</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+T</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if STUN messages will be logged.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Log m&emory reports</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+E</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if reports concerning memory management will be logged.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>System tray</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Create &system tray icon on startup</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Enable this option if you want a system tray icon for Twinkle. The system tray icon is created when you start Twinkle.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Hide in system tray when closing main window</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+H</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Enable this option if you want Twinkle to hide in the system tray when you close the main window.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Startup</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>S&tartup hidden in system tray</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Next time you start Twinkle it will immediately hide in the system tray. This works best when you also select a default user profile.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Default user profiles</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Services</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call &waiting</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+W</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Hang up &both lines when ending a 3-way conference call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+B</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Maximum calls in call history:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The maximum number of calls that will be kept in the call history.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Auto show main window on incoming call after</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+A</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>When the main window is hidden, it will be automatically shown on an incoming call after the number of specified seconds.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Number of seconds after which the main window should be shown.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>secs</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Maximum allowed size (0-65535) in bytes of an incoming SIP message over UDP.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&SIP port:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&RTP port:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Max. SIP message size (&TCP):</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The UDP/TCP port used for sending and receiving SIP messages.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Max. SIP message size (&UDP):</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Maximum allowed size (0-4294967295) in bytes of an incoming SIP message over TCP.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Play ring tone on incoming call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+P</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if a ring tone should be played when a call comes in.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Default ring tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Play the default ring tone when a call comes in.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>C&ustom ring tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+U</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Play a custom ring tone when a call comes in.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Specify the file name of a .wav file that you want to be played as ring tone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select ring tone file.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring back tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>P&lay ring back tone when network does not play ring back tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+L</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>D&efault ring back tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Play the default ring back tone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cu&stom ring back tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Play a custom ring back tone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Specify the file name of a .wav file that you want to be played as ring back tone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select ring back tone file.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Lookup name for incoming call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ove&rride received display name</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+R</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Lookup &photo for incoming call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Lookup the photo of a caller in your address book and display it on an incoming call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring tones</source> + <comment>Description of .wav files in file dialog</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose ring tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring back tones</source> + <comment>Description of .wav files in file dialog</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose ring back tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>W&eb browser command:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Command to start your web browser. If you leave this field empty Twinkle will try to figure out your default web browser.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>SysTrayPopup</name> + <message> + <source>Answer</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Reject</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Incoming Call</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>TermCapForm</name> + <message> + <source>Twinkle - Terminal Capabilities</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&From:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Get terminal capabilities of</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&To:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select an address from the address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F10</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>TransferForm</name> + <message> + <source>Twinkle - Transfer</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transfer call to</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&To:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address book</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select an address from the address book.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Type of transfer</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Blind transfer</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+B</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transfer the call to a third party without contacting that third party yourself.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>T&ransfer with consultation</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+R</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Before transferring the call to a third party, first consult the party yourself.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transfer to other &line</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+L</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Connect the remote party on the active line with the remote party on the other line.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>F10</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>TwinkleCore</name> + <message> + <source>Anonymous</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Warning:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to create log file %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot open file for reading: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>File system error while reading file %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot open file for writing: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>File system error while writing file %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Excessive number of socket errors.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Built with support for:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Contributions:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>This software contains the following software from 3rd parties:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>* GSM codec from Jutta Degener and Carsten Bormann, University of Berlin</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>* G.711/G.726 codecs from Sun Microsystems (public domain)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>* iLBC implementation from RFC 3951 (www.ilbcfreeware.org)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>* Parts of the STUN project at http://sourceforge.net/projects/stun</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>* Parts of libsrv at http://libsrv.sourceforge.net/</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For RTP the following dynamic libraries are linked:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Translated to english by <your name></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Directory %1 does not exist.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot open file %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1 is not set to your home directory.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Directory %1 (%2) does not exist.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot create directory %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1 is already running. +Lock file %2 already exists.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot create %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Syntax error in file %1 .</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to backup %1 to %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>unknown name (device is busy)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Default device</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot access the ring tone device (%1).</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot access the speaker (%1).</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot access the microphone (%1).</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot receive incoming TCP connections.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call transfer - %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Sound card cannot be set to full duplex.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot set buffer size on sound card.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Sound card cannot be set to %1 channels.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot set sound card to 16 bits recording.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot set sound card to 16 bits playing.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot set sound card sample rate to %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Opening ALSA driver failed</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot open ALSA driver for PCM playback</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot open ALSA driver for PCM capture</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot resolve STUN server: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>public IP: %1 --> private IP: %2 (SIP signaling)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>public IP: %1-%2 --> private IP: %3-%4 (RTP/RTCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot reach the STUN server: %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>If you are behind a firewall then you need to open the following UDP ports.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Port %1 (SIP signaling)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ports %1-%2 (RTP/RTCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>NAT type discovery via STUN failed.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to create file %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to write data to file %1</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Failed to send message.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Cannot lock %1 .</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>UserProfileForm</name> + <message> + <source>Twinkle - User Profile</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User profile:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select which profile you want to edit.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP server</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Voice mail</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Instant message</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Presence</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>RTP audio</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP protocol</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Transport/NAT</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Address format</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Timers</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring tones</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Scripts</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Security</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select a category for which you want to see or modify the settings.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accept and save your changes.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Undo all your changes and close the window.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP account</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&User name*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Domain*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Or&ganization:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You may fill in the name of your organization. When you make a call, this might be shown to the called party.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Your name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP authentication</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Realm:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Authentication &name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Password:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your password for authentication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Registrar</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Registrar:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Expiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The registration expiry time that Twinkle will request.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>seconds</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Re&gister at startup</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+G</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Add q-value to registration</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The q-value is a value between 0.000 and 1.000. A higher value means a higher priority.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Outbound Proxy</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Use outbound proxy</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+U</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Outbound &proxy:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send in-dialog requests to proxy</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+S</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Don't send a request to proxy if its destination can be resolved locally.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+D</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The hostname, domain name or IP address of your outbound proxy.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Co&decs</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Codecs</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Available codecs:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.711 A-law</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.711 u-law</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>GSM</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>speex-nb (8 kHz)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>speex-wb (16 kHz)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>speex-uwb (32 kHz)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>List of available codecs.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Move a codec from the list of available codecs to the list of active codecs.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Move a codec from the list of active codecs to the list of available codecs.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Active codecs:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Move a codec upwards in the list of active codecs, i.e. increase its preference of use.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Move a codec downwards in the list of active codecs, i.e. decrease its preference of use.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&G.711/G.726 payload size:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The preferred payload size for the G.711 and G.726 codecs.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>ms</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Follow codec preference from far end on incoming calls</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+F</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Follow codec &preference from far end on outgoing calls</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+P</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&iLBC</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>iLBC</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>i&LBC payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>iLBC &payload size (ms):</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for iLBC.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>20</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>30</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The preferred payload size for iLBC.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Speex</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Speex</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Perceptual &enhancement</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+E</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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).</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Ultra wide band payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+V</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Wide band payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+B</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for speex wide band.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Co&mplexity:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Discontinuous transmission is an addition to VAD/VBR operation, that allows to stop transmitting completely when the background noise is stationary.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for speex narrow band.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Narrow band payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.726</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.726 &40 kbps payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for G.726 40 kbps.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for G.726 32 kbps.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.726 &24 kbps payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for G.726 24 kbps.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.726 &32 kbps payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for G.726 16 kbps.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>G.726 &16 kbps payload type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Codeword &packing order:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>RFC 3551</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>ATM AAL2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DT&MF</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DTMF</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The dynamic type value (96 or higher) to be used for DTMF events (RFC 2833).</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DTMF vo&lume:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The power level of the DTMF tone in dB.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The pause after a DTMF tone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DTMF &duration:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DTMF payload &type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DTMF &pause:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>dB</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Duration of a DTMF tone.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>DTMF t&ransport:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Auto</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>RFC 2833</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Inband</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Out-of-band (SIP INFO)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>General</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Protocol options</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call &Hold variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>RFC 2543</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>RFC 3264</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Allow m&issing Contact header in 200 OK on REGISTER</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+I</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max-Forwards header is mandatory</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+M</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Put &registration expiry time in contact header</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+R</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Use compact header names</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if compact header names should be used for headers that have a compact form.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Allow SDP change during call setup</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use domain &name to create a unique contact header value</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+N</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Twinkle creates a unique contact header value by combining the SIP user name and domain: +</p> +<p> +<tt>&nbsp;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>&nbsp;user@local_ip</tt> +</p> +<p> +This format is what most SIP phones use. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Encode Via, Route, Record-Route as list</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Redirection</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Allow redirection</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+A</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if Twinkle should redirect a request if a 3XX response is received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ask user &permission to redirect</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if Twinkle should ask the user before redirecting a request when a 3XX response is received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Max re&directions:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP extensions</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>disabled</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>supported</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>required</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>preferred</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates 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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&100 rel (PRACK):</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Replaces</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extenstion is supported.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>REFER</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call transfer (REFER)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+T</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if Twinkle should transfer a call if a REFER request is received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>As&k user permission to transfer</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+K</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if Twinkle should ask the user before transferring a call when a REFER request is received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Hold call &with referrer while setting up call to transfer target</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+W</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if Twinkle should put the current call on hold when a REFER request to transfer a call is received.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ho&ld call with referee before sending REFER</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+L</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if Twinkle should put the current call on hold when you transfer a call.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Auto re&fresh subscription to refer event while call transfer is not finished</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Attended refer to AoR (Address of Record)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Privacy</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Privacy options</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Preferred-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Preferred-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP transport</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>UDP</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>TCP</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>T&ransport protocol:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>UDP t&hreshold:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Messages larger than the threshold are sent via TCP. Smaller messages are sent via UDP.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>NAT traversal</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&NAT traversal not needed</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose this option when there is no NAT device between you and your SIP proxy or when your SIP provider offers hosted NAT traversal.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Use statically configured public IP address inside SIP messages</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.<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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use &STUN (does not work for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose this option when your SIP provider offers a STUN server for NAT traversal.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>S&TUN server:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The hostname, domain name or IP address of the STUN server.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Public IP address:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The public IP address of your NAT.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Telephone numbers</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Only &display user part of URI for telephone number</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&URI with numerical user part is a telephone number</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Remove special symbols from numerical dial strings</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Special symbols:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The special symbols that may be part of a telephone number for nice formatting, but must be removed when dialing.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Number conversion</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Match expression</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Replace</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +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. +</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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Move the selected number conversion rule upwards in the list.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Move the selected number conversion rule downwards in the list.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Add</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Add a number conversion rule.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Re&move</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Remove the selected number conversion rule.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Edit</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Edit the selected number conversion rule.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Type a telephone number here an press the Test button to see how it is converted by the list of number conversion rules.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Test</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Test how a number is converted by the number conversion rules.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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".</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>NAT &keep alive:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&No answer:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select ring back tone file.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select ring tone file.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring &back tone:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Ring tone:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Select script file.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +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. +</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 outputing 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=&lt;string&gt;<br> +contact=&lt;address to redirect to&gt;<br> +caller_name=&lt;name of caller to display&gt;<br> +ringtone=&lt;file name of .wav file&gt;<br> +display_msg=&lt;message to show on display&gt;<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_&lt;HEADER_NAME&gt;</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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call released locall&y:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Outgoing call a&nswered:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Incoming call &failed:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Incoming call:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call released &remotely:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Incoming call &answered:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>O&utgoing call:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Out&going call failed:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Enable ZRTP/SRTP encryption</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>ZRTP settings</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>O&nly encrypt audio if remote party indicated ZRTP support in SDP</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Indicate ZRTP support in SDP</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Twinkle will indicate ZRTP support during call setup in its signalling.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Popup warning when remote party disables encryption during call</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Voice mail address:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The SIP address or telephone number to access your voice mail.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsollicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Sollicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><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>Unsollicited</H3> +<p> +Asterisk provides unsollicited message waiting indication. +</p> +<H3>Sollicited</H3> +<p> +Sollicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&MWI type:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Sollicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Subscription &duration:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Mailbox &user name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The hostname, domain name or IP address of your voice mailbox server.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your user name for accessing your voice mailbox.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Mailbox &server:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Via outbound &proxy</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Check this option if Twinkle should send SIP messages to the mailbox server via the outbound proxy.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Maximum number of sessions:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>When you have this number of instant message sessions open, new incoming message sessions will be rejected.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your presence</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Publish availability at startup</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Publish your availability at startup.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Publication &refresh interval (sec):</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Refresh rate of presence publications.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Buddy presence</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Subscription refresh interval (sec):</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Refresh rate of presence subscriptions.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Dynamic payload type %1 is used more than once.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a user name for your SIP account.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid domain.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid user name.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid value for registrar.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid value for outbound proxy.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a mailbox user name.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a mailbox server</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid mailbox server.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid mailbox user name.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Value for public IP address missing.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid value for STUN server.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring tones</source> + <comment>Description of .wav files in file dialog</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose ring tone</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Ring back tones</source> + <comment>Description of .wav files in file dialog</comment> + <translation type="unfinished"></translation> + </message> + <message> + <source>All files</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose incoming call script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose incoming call answered script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose incoming call failed script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose outgoing call script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose outgoing call answered script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose outgoing call failed script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose local release script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Choose remote release script</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>%1 converts to %2</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>P&ersistent TCP connection</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send composing indications when typing a message.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Twinkle sends a composing indication when you type a message. This way the recipient can see that you are typing.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>AKA AM&F:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>A&KA OP:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Authentication management field for AKAv1-MD5 authentication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Operator variant key for AKAv1-MD5 authentication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Prepr&ocessing</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Preprocessing (improves quality at remote end)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Automatic gain control</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Automatic gain control &level:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Automatic gain control level represents percentual value of automatic gain setting of a microphone. Recommended value is about 25%.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Voice activity detection</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>When enabled, voice activity detection detects whether the input signal represents a speech or a silence/background noise.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Noise reduction</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The noise reduction can be used to reduce the amount of background noise present in the input signal. This provides higher quality speech.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Acoustic &Echo Cancellation</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Variable &bit-rate</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Discontinuous &Transmission</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Quality:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>bytes</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use tel-URI for telephone &number</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Expand a dialed telephone number to a tel-URI instead of a sip-URI.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Accept call &transfer request (incoming REFER)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Allow call transfer while consultation in progress</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Enable NAT &keep alive</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Send UDP NAT keep alive packets.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>WizardForm</name> + <message> + <source>Twinkle - Wizard</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>The hostname, domain name or IP address of the STUN server.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>S&TUN server:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Domain*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Authentication name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Your name:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>SIP pro&xy:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&SIP service provider:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Password:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&User name*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Your password for authentication.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&OK</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+O</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Cancel</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Alt+C</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>None (direct IP to IP calls)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Other</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>User profile wizard:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>You must fill in a user name for your SIP account.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>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.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid value for SIP proxy.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Invalid value for STUN server.</source> + <translation type="unfinished"></translation> + </message> +</context> +<context> + <name>YesNoDialog</name> + <message> + <source>&Yes</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&No</source> + <translation type="unfinished"></translation> + </message> +</context> +</TS> |