diff options
Diffstat (limited to 'src')
| -rw-r--r-- | src/gui/lang/twinkle_cs.ts | 153 | ||||
| -rw-r--r-- | src/gui/lang/twinkle_de.ts | 4 | ||||
| -rw-r--r-- | src/gui/lang/twinkle_nl.ts | 155 | ||||
| -rw-r--r-- | src/gui/lang/twinkle_ru.ts | 161 | ||||
| -rw-r--r-- | src/gui/lang/twinkle_sk.ts | 159 | ||||
| -rw-r--r-- | src/gui/lang/twinkle_sv.ts | 222 | ||||
| -rw-r--r-- | src/gui/lang/twinkle_xx.ts | 228 |
7 files changed, 808 insertions, 274 deletions
diff --git a/src/gui/lang/twinkle_cs.ts b/src/gui/lang/twinkle_cs.ts index 34586ae..9be2844 100644 --- a/src/gui/lang/twinkle_cs.ts +++ b/src/gui/lang/twinkle_cs.ts @@ -955,6 +955,14 @@ <translation>Zdá se, že <p><b>KAddressbook</b> neobsahuje žádné záznamy s telefonními čísly, které by Twinkle mohl načíst. Použijte prosím tento program k úpravě nebo zanesení vašich kontaktů.</p> <p>Druhou možností je používat místní adresář v Twinkle.</p></translation> </message> + <message> + <source>Are you sure you want to delete contact '%1' from the local address book?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete contact</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>GetProfileNameForm</name> @@ -4155,11 +4163,11 @@ a na vašem NATu namapovat (UDP) porty.</translation> </message> <message> <source>&Domain*:</source> - <translation>&Doména*:</translation> + <translation type="vanished">&Doména*:</translation> </message> <message> <source>Or&ganization:</source> - <translation>Or&ganizace:</translation> + <translation type="vanished">Or&ganizace:</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. @@ -4233,7 +4241,7 @@ Toto pole je povinné.</translation> </message> <message> <source>&Expiry:</source> - <translation>&Platnost:</translation> + <translation type="vanished">&Platnost:</translation> </message> <message> <source>The registration expiry time that Twinkle will request.</source> @@ -4635,7 +4643,7 @@ Vysílá DTMF out-of-band přes požadavek SIP INFO.</p></translation> </message> <message> <source>Max re&directions:</source> - <translation>Max. počet &přesměrování:</translation> + <translation type="vanished">Max. počet &přesměrování:</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> @@ -4647,7 +4655,7 @@ Vysílá DTMF out-of-band přes požadavek SIP INFO.</p></translation> </message> <message> <source>Call &Hold variant:</source> - <translation>Způsob přidržení &hovoru:</translation> + <translation type="vanished">Způsob přidržení &hovoru:</translation> </message> <message> <source>RFC 2543</source> @@ -4904,7 +4912,7 @@ Pokud si tuto volbu vyberete, musíte rovněž na vašem NAT zařízení nasměr </message> <message> <source>S&TUN server:</source> - <translation>Adresa S&TUN serveru:</translation> + <translation type="vanished">Adresa S&TUN serveru:</translation> </message> <message> <source>The hostname, domain name or IP address of the STUN server.</source> @@ -4998,7 +5006,7 @@ You are at work and all telephone numbers starting with a 0 should be prefixed w Match expression = 0[0-9]* , Replace = 9$&<br> </tt> </blockquote></source> - <translation><p> + <translation type="vanished"><p> Často není formát telefonních čísel, které jsou očekávány od VoIP poskytovatele, shodný s formátem čísel uložených v adresáři. Např. u čísel začínajících na "+" a národním kódem země očekává váš poskytovatel namísto "00" znak "+". Nebo jste-li napojeni na místní SIP síť a je nutné předtočit nejdříve číslo k přístupu ven. Zde je možné za použití vyhledávacích a zaměňovacích vzorů (podle způsobu regulárních výrazů a la Perl) nastavit obecně platné pravidla pro konverzi telefonních čísel. </p> @@ -5326,7 +5334,7 @@ Obsahy všech SIP hlaviček odeslaných SIP INVITE požadavků budou předány t </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>Pokud je aktivováno, pokusí se Twinkle při všech odchozích a příchozích hovorech zašifrovat zvuková data. Aby byl hovor opravdu zašifrován musí i protistrana podporovat šifrování ZRTP/SRTP. Jinak zůstane hovor nešifrovaný.</translation> + <translation type="vanished">Pokud je aktivováno, pokusí se Twinkle při všech odchozích a příchozích hovorech zašifrovat zvuková data. Aby byl hovor opravdu zašifrován musí i protistrana podporovat šifrování ZRTP/SRTP. Jinak zůstane hovor nešifrovaný.</translation> </message> <message> <source>ZRTP settings</source> @@ -5494,7 +5502,7 @@ Pokud je deaktivováno, použije Twinkle první kodek z vlastního seznamu, kter </message> <message> <source>Indicates if the Replaces-extenstion is supported.</source> - <translation>Indikuje, zda je rozšíření Replaces podporováno.</translation> + <translation type="vanished">Indikuje, zda je rozšíření Replaces podporováno.</translation> </message> <message> <source>Attended refer to AoR (Address of Record)</source> @@ -5502,7 +5510,7 @@ Pokud je deaktivováno, použije Twinkle první kodek z vlastního seznamu, kter </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>Asistované přepojení by mělo používat Contact-URI jako cílovou adresu pro sdělení nového spojení přesměrovávané protistraně. Tato adresa nemusí být ovšem globálně platná. Alternativně se může použít AoR (Address of Record). Nevýhodou je, že při více koncových zařízeních není AoR jednoznačné, zatímco URI kontaktu směřuje na jediné zařízení.</translation> + <translation type="vanished">Asistované přepojení by mělo používat Contact-URI jako cílovou adresu pro sdělení nového spojení přesměrovávané protistraně. Tato adresa nemusí být ovšem globálně platná. Alternativně se může použít AoR (Address of Record). Nevýhodou je, že při více koncových zařízeních není AoR jednoznačné, zatímco URI kontaktu směřuje na jediné zařízení.</translation> </message> <message> <source>Privacy</source> @@ -5616,11 +5624,11 @@ TWINKLE_USER_PROFILE obsahuje jméno uživatelského profilu, pro který je př </message> <message> <source>Unsollicited</source> - <translation>Nevyžádané</translation> + <translation type="vanished">Nevyžádané</translation> </message> <message> <source>Sollicited</source> - <translation>Vyžádané</translation> + <translation type="vanished">Vyžádané</translation> </message> <message> <source><H2>Message waiting indication type</H2> @@ -5635,7 +5643,7 @@ Asterisk provides unsollicited message waiting indication. <p> Sollicited message waiting indication as specified by RFC 3842. </p></source> - <translation><H2>Typ indikace čekajících zpráv</H2> + <translation type="vanished"><H2>Typ indikace čekajících zpráv</H2> <p> Pokud váš SIP poskytovatel nabízí upozornění na uložené zprávy v hlasové schránce, může vás Twinkle informovat o nových i již vyslechnutých zprávách ve vaší hlasové schránce. Zeptejte se vašeho poskytovatele, jaký typ indikace čekajících zpráv je používán </p> @@ -5654,7 +5662,7 @@ Vyžádaná indikace čekajících zpráv dle RFC 3842. </message> <message> <source>Sollicited MWI</source> - <translation>Vyžádané MWI</translation> + <translation type="vanished">Vyžádané MWI</translation> </message> <message> <source>Subscription &duration:</source> @@ -5670,7 +5678,7 @@ Vyžádaná indikace čekajících zpráv dle RFC 3842. </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>Dle specifikace MWI se koncové zařízení hlásí na serveru k příjmu zpráv na určitou dobu a před vypršením této doby by se přihlášení mělo znovu obnovit.</translation> + <translation type="vanished">Dle specifikace MWI se koncové zařízení hlásí na serveru k příjmu zpráv na určitou dobu a před vypršením této doby by se přihlášení mělo znovu obnovit.</translation> </message> <message> <source>Your user name for accessing your voice mailbox.</source> @@ -5818,7 +5826,7 @@ Vyžádaná indikace čekajících zpráv dle RFC 3842. </message> <message> <source>Use &STUN (does not work for incoming TCP)</source> - <translation>Použít &STUN (nefunguje pro příchozí TCP)</translation> + <translation type="vanished">Použít &STUN (nefunguje pro příchozí TCP)</translation> </message> <message> <source>P&ersistent TCP connection</source> @@ -5952,6 +5960,119 @@ Vyžádaná indikace čekajících zpráv dle RFC 3842. <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>Pokud máte povolený STUN nebo NAT keep alive, pak bude Twinkle zasílat udržovací pakety v tomto intervalu, aby byla udržena mapování na vašem NATu.</translation> </message> + <message> + <source>Do&main*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Organi&zation:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>E&xpiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call Hold &variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max redirections:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extension is supported.</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 endpoint.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Asserted-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use STUN (does not wor&k for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>STUN ser&ver:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +</p> +<p> +For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. +</p> +<p> +The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. +</p> +<h3>Example 1</h3> +<p> +Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. +</p> +<p> +The following rules will do the trick: +</p> +<blockquote> +<tt> +Match expression = \+31([0-9]*) , Replace = 0$1<br> +Match expression = \+([0-9]*) , Replace = 00$1</br> +</tt> +</blockquote> +<h3>Example 2</h3> +<p> +You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. +</p> +<blockquote> +<tt> +Match expression = 0[0-9]* , Replace = 9$&<br> +</tt> +</blockquote></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 unencrypted.</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>Unsolicited</H3> +<p> +Asterisk provides unsolicited message waiting indication. +</p> +<H3>Solicited</H3> +<p> +Solicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsolicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For solicited 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> </context> <context> <name>WizardForm</name> diff --git a/src/gui/lang/twinkle_de.ts b/src/gui/lang/twinkle_de.ts index 8ab0b01..8adc278 100644 --- a/src/gui/lang/twinkle_de.ts +++ b/src/gui/lang/twinkle_de.ts @@ -1,6 +1,6 @@ <?xml version="1.0" encoding="utf-8"?> <!DOCTYPE TS> -<TS version="2.0"> +<TS version="2.1" language="de" sourcelanguage="en"> <context> <name>AddressCardForm</name> <message> @@ -2419,7 +2419,7 @@ Um den Online-Status eines Buddies abzufragen, muss <i>dessen</i> Pr </message> <message> <source>Reject</source> - <translation >Abweisen</translation> + <translation>Abweisen</translation> </message> <message> <source>&Hold</source> diff --git a/src/gui/lang/twinkle_nl.ts b/src/gui/lang/twinkle_nl.ts index 131094e..adbf748 100644 --- a/src/gui/lang/twinkle_nl.ts +++ b/src/gui/lang/twinkle_nl.ts @@ -1,6 +1,6 @@ <?xml version="1.0" encoding="utf-8"?> <!DOCTYPE TS> -<TS version="2.0"> +<TS version="2.1" language="nl" sourcelanguage="en"> <context> <name>AddressCardForm</name> <message> @@ -1003,6 +1003,14 @@ <translation><p>U heeft geen contacten met een telefoonnummer in <b>KAddressBook</b>, KDE's adresboek applicatie. Twinkle haalt alle contacten met een telefoonnummer uit KAdressBook. Om uw contacten te beheren, moet u KAddressbook gebruiken.</p> <p>Als alternatief kunt u het lokale adresboek van Twinkle gebruiken.</p></translation> </message> + <message> + <source>Are you sure you want to delete contact '%1' from the local address book?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete contact</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>GetProfileNameForm</name> @@ -4237,11 +4245,11 @@ en creëer de volgende statische UDP mapping in uw NAT.</translation> </message> <message> <source>&Domain*:</source> - <translation>&Domein*:</translation> + <translation type="vanished">&Domein*:</translation> </message> <message> <source>Or&ganization:</source> - <translation>Or&ganisatie:</translation> + <translation type="vanished">Or&ganisatie:</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. @@ -4309,7 +4317,7 @@ Dit is een verplicht veld.</translation> </message> <message> <source>&Expiry:</source> - <translation>&Interval:</translation> + <translation type="vanished">&Interval:</translation> </message> <message> <source>The registration expiry time that Twinkle will request.</source> @@ -4721,7 +4729,7 @@ Stuur DTMF out-of-band in een SIP INFO verzoek. </message> <message> <source>Max re&directions:</source> - <translation>Max &doorverwijzingen:</translation> + <translation type="vanished">Max &doorverwijzingen:</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> @@ -4733,7 +4741,7 @@ Stuur DTMF out-of-band in een SIP INFO verzoek. </message> <message> <source>Call &Hold variant:</source> - <translation>Wac&ht variant:</translation> + <translation type="vanished">Wac&ht variant:</translation> </message> <message> <source>RFC 2543</source> @@ -4990,7 +4998,7 @@ Als u deze optie kiest, dan moet u teven een adres vertaling in uw NAT router aa </message> <message> <source>S&TUN server:</source> - <translation>S&TUN server:</translation> + <translation type="vanished">S&TUN server:</translation> </message> <message> <source>The hostname, domain name or IP address of the STUN server.</source> @@ -5084,7 +5092,7 @@ You are at work and all telephone numbers starting with a 0 should be prefixed w Match expression = 0[0-9]* , Replace = 9$&<br> </tt> </blockquote></source> - <translation><p> + <translation type="vanished"><p> Vaak is het formaat van een telefoonnummer dat u moet draaien anders dan het formaat van het nummer in uw adresboek, bijv. uw nummers starten met een +-teken gevolgd door een landencode, maar uw provider verwacht '00' in plaats van het +-teken, of u bent op kantoor en u moet eerst een '9' draaien om naar buiten te bellen. Hier kunt u nummerformaatconversie definieren m.b.v. reguliere expressies en vervang strings (Perl syntax). </p> <p> @@ -5390,7 +5398,7 @@ De waarden van alle SIP headers van de uitgaande SIP INVITE worden via variabele </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>Als u ZRTP/SRTP aanzet, dan zal Twinkle proberen om het audiokanaal van uw gesprekken te versleutelen. Versleuteling lukt alleen als uw gesprekspartner ook ZRTP/SRTP ondersteunt. Als uw gesprekspartner geen ZRTP/SRTP ondersteund, dan blijft het audiokanaal onversleuteld.</translation> + <translation type="vanished">Als u ZRTP/SRTP aanzet, dan zal Twinkle proberen om het audiokanaal van uw gesprekken te versleutelen. Versleuteling lukt alleen als uw gesprekspartner ook ZRTP/SRTP ondersteunt. Als uw gesprekspartner geen ZRTP/SRTP ondersteund, dan blijft het audiokanaal onversleuteld.</translation> </message> <message> <source>ZRTP settings</source> @@ -5558,7 +5566,7 @@ Als u deze optie uitschakelt, dan neemt Twinkle de eerste codec uit de actieve c </message> <message> <source>Indicates if the Replaces-extenstion is supported.</source> - <translation>Geeft aan of de Replaces-extensie ondersteund wordt.</translation> + <translation type="vanished">Geeft aan of de Replaces-extensie ondersteund wordt.</translation> </message> <message> <source>Attended refer to AoR (Address of Record)</source> @@ -5566,7 +5574,7 @@ Als u deze optie uitschakelt, dan neemt Twinkle de eerste codec uit de actieve c </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>Bij begeleid doorverbinden, is de contact URI de doorverbindbestemming. Een contact URI kan echter niet globaal routeerbaar zijn. Als alternatief kan dan de AoR (Address of Record) gebruikt worden. Een nadeel van het gebruik van de AoR is dat deze routeerbaar kan zijn naar meerdere eindpunten in het geval van SIP forking. De contact URI routeert altijd naar een uniek eindpunt.</translation> + <translation type="vanished">Bij begeleid doorverbinden, is de contact URI de doorverbindbestemming. Een contact URI kan echter niet globaal routeerbaar zijn. Als alternatief kan dan de AoR (Address of Record) gebruikt worden. Een nadeel van het gebruik van de AoR is dat deze routeerbaar kan zijn naar meerdere eindpunten in het geval van SIP forking. De contact URI routeert altijd naar een uniek eindpunt.</translation> </message> <message> <source>Privacy</source> @@ -5700,11 +5708,11 @@ TWINKLE_TRIGGER=in_call. SIPREQUEST_METHOD=INVITE. <b>SIPREQUEST_URI</b </message> <message> <source>Unsollicited</source> - <translation>Unsollicited</translation> + <translation type="vanished">Unsollicited</translation> </message> <message> <source>Sollicited</source> - <translation>Sollicited</translation> + <translation type="vanished">Sollicited</translation> </message> <message> <source><H2>Message waiting indication type</H2> @@ -5719,7 +5727,7 @@ Asterisk provides unsollicited message waiting indication. <p> Sollicited message waiting indication as specified by RFC 3842. </p></source> - <translation><H2>Message waiting indication type</H2> + <translation type="vanished"><H2>Message waiting indication type</H2> <p> Als uw provider de dienst aanbiedt waarmee u uw voice mail status kunt zien, dan kan Twinkle laten zien hoeveel nieuwe voice mail berichten er op u wachten. Er zijn 2 methoden waarop deze dienst kan worden aangeboden. </p> @@ -5738,7 +5746,7 @@ Sollicited message waiting indication zoals gespecificeerd in RFC 3842. </message> <message> <source>Sollicited MWI</source> - <translation>Sollicited MWI</translation> + <translation type="vanished">Sollicited MWI</translation> </message> <message> <source>Subscription &duration:</source> @@ -5754,7 +5762,7 @@ Sollicited message waiting indication zoals gespecificeerd in RFC 3842. </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>Twinkle meldt zich voor een bepaalde periode aan bij de voice mailbox server. Net voordat deze periode verstrijkt, zal Twinkle zich opnieuw aanmelden.</translation> + <translation type="vanished">Twinkle meldt zich voor een bepaalde periode aan bij de voice mailbox server. Net voordat deze periode verstrijkt, zal Twinkle zich opnieuw aanmelden.</translation> </message> <message> <source>Your user name for accessing your voice mailbox.</source> @@ -5922,7 +5930,7 @@ Sollicited message waiting indication zoals gespecificeerd in RFC 3842. </message> <message> <source>Use &STUN (does not work for incoming TCP)</source> - <translation>&STUN (werkt niet voor inkomend TCP verkeer)</translation> + <translation type="vanished">&STUN (werkt niet voor inkomend TCP verkeer)</translation> </message> <message> <source>P&ersistent TCP connection</source> @@ -6056,6 +6064,119 @@ Sollicited message waiting indication zoals gespecificeerd in RFC 3842. <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>Als u STUN of NAT keep alive aan heeft gezet, dan zal Twinkle keep alive pakketjes sturen met deze snelheid om de adresbindingen in uw NAT router in leven te houden.</translation> </message> + <message> + <source>Do&main*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Organi&zation:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>E&xpiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call Hold &variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max redirections:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extension is supported.</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 endpoint.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Asserted-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use STUN (does not wor&k for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>STUN ser&ver:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +</p> +<p> +For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. +</p> +<p> +The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. +</p> +<h3>Example 1</h3> +<p> +Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. +</p> +<p> +The following rules will do the trick: +</p> +<blockquote> +<tt> +Match expression = \+31([0-9]*) , Replace = 0$1<br> +Match expression = \+([0-9]*) , Replace = 00$1</br> +</tt> +</blockquote> +<h3>Example 2</h3> +<p> +You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. +</p> +<blockquote> +<tt> +Match expression = 0[0-9]* , Replace = 9$&<br> +</tt> +</blockquote></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 unencrypted.</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>Unsolicited</H3> +<p> +Asterisk provides unsolicited message waiting indication. +</p> +<H3>Solicited</H3> +<p> +Solicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsolicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For solicited 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> </context> <context> <name>WizardForm</name> diff --git a/src/gui/lang/twinkle_ru.ts b/src/gui/lang/twinkle_ru.ts index 8b60aa9..2751999 100644 --- a/src/gui/lang/twinkle_ru.ts +++ b/src/gui/lang/twinkle_ru.ts @@ -1,4 +1,6 @@ -<?xml version="1.0" ?><!DOCTYPE TS><TS language="ru" version="2.0"> +<?xml version="1.0" encoding="utf-8"?> +<!DOCTYPE TS> +<TS version="2.1" language="ru"> <context> <name>AddressCardForm</name> <message> @@ -917,6 +919,14 @@ <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><p>У Вас нет ни одного контакта с телефонным номером в <b>KAddressBook</b>(приложении KDE адресная книга). Twinkle получает все контакты с телефонными номерами из Адресной книги KDE. Для управления вашими контактами вы должны использовать KAddressBook.<p>Как альтернативу вы можете использовать локальную адресную книгу Twinkle.</p></translation> </message> + <message> + <source>Are you sure you want to delete contact '%1' from the local address book?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete contact</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>GetProfileNameForm</name> @@ -3891,11 +3901,11 @@ STUN не работает. </message> <message> <source>&Domain*:</source> - <translation>&Домен*:</translation> + <translation type="vanished">&Домен*:</translation> </message> <message> <source>Or&ganization:</source> - <translation>Ор&ганизация:</translation> + <translation type="vanished">Ор&ганизация:</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. @@ -3967,7 +3977,7 @@ This field is mandatory.</source> </message> <message> <source>&Expiry:</source> - <translation>&Устаревание:</translation> + <translation type="vanished">&Устаревание:</translation> </message> <message> <source>The registration expiry time that Twinkle will request.</source> @@ -4406,7 +4416,7 @@ Send DTMF out-of-band via a SIP INFO request. </message> <message> <source>Call &Hold variant:</source> - <translation>&Вариант удержания вызова:</translation> + <translation type="vanished">&Вариант удержания вызова:</translation> </message> <message> <source>RFC 2543</source> @@ -4553,7 +4563,7 @@ This format is what most SIP phones use. </message> <message> <source>Max re&directions:</source> - <translation>&Максимум перенаправлений:</translation> + <translation type="vanished">&Максимум перенаправлений:</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> @@ -4607,7 +4617,7 @@ This format is what most SIP phones use. </message> <message> <source>Indicates if the Replaces-extenstion is supported.</source> - <translation>Указывает, поддерживается ли расширение Replaces.</translation> + <translation type="vanished">Указывает, поддерживается ли расширение Replaces.</translation> </message> <message> <source>REFER</source> @@ -4675,7 +4685,7 @@ This format is what most SIP phones use. </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>Приходящая передача вызова должна использовать URI контакта в качестве адресата для связи нового соединения, перенаправленного на контрагента. Однако этот адрес не может быть глобально действительным. В качестве альтернативы можно использовать AoR (адрес записи). Недостатком является то, что с несколькими конечными устройствами AoR однозначен, тогда как URI контакта направляется на одно устройство.</translation> + <translation type="vanished">Приходящая передача вызова должна использовать URI контакта в качестве адресата для связи нового соединения, перенаправленного на контрагента. Однако этот адрес не может быть глобально действительным. В качестве альтернативы можно использовать AoR (адрес записи). Недостатком является то, что с несколькими конечными устройствами AoR однозначен, тогда как URI контакта направляется на одно устройство.</translation> </message> <message> <source>Privacy</source> @@ -4745,7 +4755,7 @@ When you choose this option you have to create static address mappings in your N </message> <message> <source>Use &STUN (does not work for incoming TCP)</source> - <translation>Использовать &STUN (не работает для входящего TCP)</translation> + <translation type="vanished">Использовать &STUN (не работает для входящего TCP)</translation> </message> <message> <source>Choose this option when your SIP provider offers a STUN server for NAT traversal.</source> @@ -4753,7 +4763,7 @@ When you choose this option you have to create static address mappings in your N </message> <message> <source>S&TUN server:</source> - <translation>S&TUN сервер:</translation> + <translation type="vanished">S&TUN сервер:</translation> </message> <message> <source>The hostname, domain name or IP address of the STUN server.</source> @@ -4817,7 +4827,7 @@ When you choose this option you have to create static address mappings in your N </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. +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. @@ -4827,7 +4837,7 @@ The number conversion rules are also applied to incoming calls, so the numbers a </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'. +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: @@ -4847,7 +4857,7 @@ You are at work and all telephone numbers starting with a 0 should be prefixed w Match expression = 0[0-9]* , Replace = 9$&<br> </tt> </blockquote></source> - <translation><p> + <translation type="vanished"><p> Часто количество телефонных номеров, ожидаемых от провайдера VoIP, не совпадает с количеством, сохранённым в каталоге. К примеру, для номеров, начинающихся с «+» и национального кода страны, ваш провайдер ожидает «+» вместо «00». Или, если вы подключены к локальной сети SIP, вам необходимо предварительно указать номер доступа. Можно установить общепринятые правила для преобразования телефонных номеров с использованием шаблонов поиска и свопинга (с помощью регулярных выражений и языка Perl). </p> @@ -5291,7 +5301,7 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v </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>Если включено, Twinkle пытается зашифровать аудиоданные во всех исходящих и входящих вызовах. Для того, чтобы вызов был зашифрован, контрагент должен также поддерживать шифрование ZRTP / SRTP. В противном случае вызов остаётся незашифрованным.</translation> + <translation type="vanished">Если включено, Twinkle пытается зашифровать аудиоданные во всех исходящих и входящих вызовах. Для того, чтобы вызов был зашифрован, контрагент должен также поддерживать шифрование ZRTP / SRTP. В противном случае вызов остаётся незашифрованным.</translation> </message> <message> <source>ZRTP settings</source> @@ -5331,11 +5341,11 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v </message> <message> <source>Unsollicited</source> - <translation>Незапрошен</translation> + <translation type="vanished">Незапрошен</translation> </message> <message> <source>Sollicited</source> - <translation>Запрошен</translation> + <translation type="vanished">Запрошен</translation> </message> <message> <source><H2>Message waiting indication type</H2> @@ -5350,7 +5360,7 @@ Asterisk provides unsollicited message waiting indication. <p> Sollicited message waiting indication as specified by RFC 3842. </p></source> - <translation><H2>Тип отображения ожидающего сообщения</H2> + <translation type="vanished"><H2>Тип отображения ожидающего сообщения</H2> <p> Если ваш SIP-провайдер предлагает оповещения о сохранённых сообщениях в вашей голосовой почте, Twinkle может рассказать вам о новых и уже услышанных сообщениях в вашей голосовой почте. Спросите своего провайдера, какой тип ожидающего сообщения используется </p> @@ -5369,7 +5379,7 @@ Asterisk поддерживает нежелательные ожидающие </message> <message> <source>Sollicited MWI</source> - <translation>Запрошенный MWI</translation> + <translation type="vanished">Запрошенный MWI</translation> </message> <message> <source>Subscription &duration:</source> @@ -5385,7 +5395,7 @@ Asterisk поддерживает нежелательные ожидающие </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>Согласно спецификации MWI, конечное устройство сообщает серверу о получении сообщения в течение определённого периода времени, и до истечения этого времени регистрация должна возобновиться.</translation> + <translation type="vanished">Согласно спецификации MWI, конечное устройство сообщает серверу о получении сообщения в течение определённого периода времени, и до истечения этого времени регистрация должна возобновиться.</translation> </message> <message> <source>Your user name for accessing your voice mailbox.</source> @@ -5683,6 +5693,119 @@ This could be the hostname or IP address of your PC if you want direct PC to PC <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>Если у вас включен STUN или NAT, то Twinkle отправит пакеты обслуживания на этот интервал, чтобы сохранить отображение на вашем NAT.</translation> </message> + <message> + <source>Do&main*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Organi&zation:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>E&xpiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call Hold &variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max redirections:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extension is supported.</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 endpoint.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Asserted-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use STUN (does not wor&k for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>STUN ser&ver:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +</p> +<p> +For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. +</p> +<p> +The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. +</p> +<h3>Example 1</h3> +<p> +Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. +</p> +<p> +The following rules will do the trick: +</p> +<blockquote> +<tt> +Match expression = \+31([0-9]*) , Replace = 0$1<br> +Match expression = \+([0-9]*) , Replace = 00$1</br> +</tt> +</blockquote> +<h3>Example 2</h3> +<p> +You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. +</p> +<blockquote> +<tt> +Match expression = 0[0-9]* , Replace = 9$&<br> +</tt> +</blockquote></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 unencrypted.</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>Unsolicited</H3> +<p> +Asterisk provides unsolicited message waiting indication. +</p> +<H3>Solicited</H3> +<p> +Solicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsolicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For solicited 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> </context> <context> <name>WizardForm</name> diff --git a/src/gui/lang/twinkle_sk.ts b/src/gui/lang/twinkle_sk.ts index 247de18..3bcbd69 100644 --- a/src/gui/lang/twinkle_sk.ts +++ b/src/gui/lang/twinkle_sk.ts @@ -939,6 +939,14 @@ <translation>Zdá sa, že <p><b>KAddressBook</b> neobsahuje žiadne záznamy s telefónnymi číslami, ktoré by Twinkle mohol načítať. Twinkle načítava z KAddressBook všetky kontakty s telefónnym číslom. Použite prosím tento program pre úpravu alebo pridanie vašich kontaktov.</p> <p>Druhou možnosťou je použiť miestny adresár v Twinkle.</p></translation> </message> + <message> + <source>Are you sure you want to delete contact '%1' from the local address book?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete contact</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>GetProfileNameForm</name> @@ -3891,11 +3899,11 @@ a na vašom NATe namapujte statické (UDP) porty.</translation> </message> <message> <source>&Domain*:</source> - <translation>&Doména*:</translation> + <translation type="vanished">&Doména*:</translation> </message> <message> <source>Or&ganization:</source> - <translation>Or&ganizácia:</translation> + <translation type="vanished">Or&ganizácia:</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. @@ -3967,7 +3975,7 @@ Toto pole je povinné.</translation> </message> <message> <source>&Expiry:</source> - <translation>&Platnosť:</translation> + <translation type="vanished">&Platnosť:</translation> </message> <message> <source>The registration expiry time that Twinkle will request.</source> @@ -4364,7 +4372,7 @@ Vysielať DTMF out-of-band prostredníctvom požiadavky SIP INFO.</p></tra </message> <message> <source>Max re&directions:</source> - <translation>Max. počet &presmerovaní:</translation> + <translation type="vanished">Max. počet &presmerovaní:</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> @@ -4376,7 +4384,7 @@ Vysielať DTMF out-of-band prostredníctvom požiadavky SIP INFO.</p></tra </message> <message> <source>Call &Hold variant:</source> - <translation>Spôsob pridržania &hovoru:</translation> + <translation type="vanished">Spôsob pridržania &hovoru:</translation> </message> <message> <source>RFC 2543</source> @@ -4625,7 +4633,7 @@ Ak si vyberiete túto voľbu, musíte tiež na vašom zariadení NAT nasmerovať </message> <message> <source>S&TUN server:</source> - <translation>Adresa S&TUN servera:</translation> + <translation type="vanished">Adresa S&TUN servera:</translation> </message> <message> <source>The hostname, domain name or IP address of the STUN server.</source> @@ -4719,7 +4727,7 @@ You are at work and all telephone numbers starting with a 0 should be prefixed w Match expression = 0[0-9]* , Replace = 9$&<br> </tt> </blockquote></source> - <translation><p> + <translation type="vanished"><p> Často sa formát telefónnych čísiel, ktoré očakáva poskytovateľ VoIP nezhoduje s formátom čísiel uložených v adresári. Napr. pri číslach začínajúcich na "+" a národným kódom krajiny očakáva váš poskytovateľ miesto "00" znak "+". Alebo ste pripojený na miestnu sieť SIP a je potrebné najprv zadať predvoľbu pre volania smerom von. Tu je možné pomocou vyhľadávacích a nahradzovaných vzorov (podľa štýlu regulárnych výrazov so syntaxom jazyku Perl) nastaviť všeobecne platné pravidlá pre konverziu telefónnych čísiel. </p> @@ -5038,7 +5046,7 @@ Obsahy všetkých SIP hlavičiek odosielaných SIP INVITE požiadaviek budú odo </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>Ak je aktivované, Twinkle sa pokúsi pri všetkých odchádzajúcich a prichádzajúcich hovoroch šifrovať zvukové dáta. Aby bol hovor naozaj zašifrovaný musí aj protistrana podporovať šifrovanie ZRTP/SRTP. Inak ostane hovor nešifrovaný.</translation> + <translation type="vanished">Ak je aktivované, Twinkle sa pokúsi pri všetkých odchádzajúcich a prichádzajúcich hovoroch šifrovať zvukové dáta. Aby bol hovor naozaj zašifrovaný musí aj protistrana podporovať šifrovanie ZRTP/SRTP. Inak ostane hovor nešifrovaný.</translation> </message> <message> <source>ZRTP settings</source> @@ -5206,7 +5214,7 @@ Ak je voľba vypnutá, použije Twinkle prvý kodek vo vlastnom zozname, ktorý </message> <message> <source>Indicates if the Replaces-extenstion is supported.</source> - <translation>Indikuje, či je podporované rozšírenie Replaces.</translation> + <translation type="vanished">Indikuje, či je podporované rozšírenie Replaces.</translation> </message> <message> <source>Attended refer to AoR (Address of Record)</source> @@ -5214,7 +5222,7 @@ Ak je voľba vypnutá, použije Twinkle prvý kodek vo vlastnom zozname, ktorý </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>Asistované prepojenie by malo používať Contact URI ako cieľovú adresu pre informovanie presmerovávanej strany o novom spojení. Táto adresa nemusí byť však globálne platiť. Ako alternatívu je možné použiť AoR (Address of Record). Nevýhodou je, že pri viacerých koncových zariadeniach nie je AoR jednoznačné, zatiaľčo URI kontaktu vždy ukazuje na jediné zariadenie.</translation> + <translation type="vanished">Asistované prepojenie by malo používať Contact URI ako cieľovú adresu pre informovanie presmerovávanej strany o novom spojení. Táto adresa nemusí byť však globálne platiť. Ako alternatívu je možné použiť AoR (Address of Record). Nevýhodou je, že pri viacerých koncových zariadeniach nie je AoR jednoznačné, zatiaľčo URI kontaktu vždy ukazuje na jediné zariadenie.</translation> </message> <message> <source>Privacy</source> @@ -5328,11 +5336,11 @@ TWINKLE_USER_PROFILE obsahuje meno používateľského profilu, pre ktorý je pr </message> <message> <source>Unsollicited</source> - <translation>Nevyžiadané</translation> + <translation type="vanished">Nevyžiadané</translation> </message> <message> <source>Sollicited</source> - <translation>Vyžiadané</translation> + <translation type="vanished">Vyžiadané</translation> </message> <message> <source><H2>Message waiting indication type</H2> @@ -5347,7 +5355,7 @@ Asterisk provides unsollicited message waiting indication. <p> Sollicited message waiting indication as specified by RFC 3842. </p></source> - <translation><H2>Typ indikácie čakajúcich správ</H2> + <translation type="vanished"><H2>Typ indikácie čakajúcich správ</H2> <p> Ak váš poskytovateľ SIP ponúka upozornenie na uložené správy v hlasovej schránke, môže vás Twinkle informovať o nových aj už vypočutých správach vo vašej hlasovej schránke. Spýtajte sa vášho poskytovateľa, aký typ indikácie čakajúcich správ je používaný @@ -5367,7 +5375,7 @@ Vyžiadaná indikácia čakajúcich správ podľa RFC 3842. </message> <message> <source>Sollicited MWI</source> - <translation>Vyžiadané MWI</translation> + <translation type="vanished">Vyžiadané MWI</translation> </message> <message> <source>Subscription &duration:</source> @@ -5383,7 +5391,7 @@ Vyžiadaná indikácia čakajúcich správ podľa RFC 3842. </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>Podľa špecifikácie MWI sa koncové zariadenie prihlási na serveri k príjmu správ na určitú dobu. Pred vypršaním tejto doby by sa prihlásenie malo znovu obnoviť.</translation> + <translation type="vanished">Podľa špecifikácie MWI sa koncové zariadenie prihlási na serveri k príjmu správ na určitú dobu. Pred vypršaním tejto doby by sa prihlásenie malo znovu obnoviť.</translation> </message> <message> <source>Your user name for accessing your voice mailbox.</source> @@ -5531,7 +5539,7 @@ Vyžiadaná indikácia čakajúcich správ podľa RFC 3842. </message> <message> <source>Use &STUN (does not work for incoming TCP)</source> - <translation>Použiť &STUN (nefunguje pre prichádzajúce TCP)</translation> + <translation type="vanished">Použiť &STUN (nefunguje pre prichádzajúce TCP)</translation> </message> <message> <source>P&ersistent TCP connection</source> @@ -5665,6 +5673,119 @@ Vyžiadaná indikácia čakajúcich správ podľa RFC 3842. <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>Ak máte povolený STUN alebo NAT keep alive, bude Twinkle zasielať udržovacie pakety v tomto intervale tak, aby boli udržané mapovania na vašom NATe.</translation> </message> + <message> + <source>Do&main*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Organi&zation:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>E&xpiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call Hold &variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max redirections:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extension is supported.</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 endpoint.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Asserted-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use STUN (does not wor&k for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>STUN ser&ver:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +</p> +<p> +For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. +</p> +<p> +The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. +</p> +<h3>Example 1</h3> +<p> +Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. +</p> +<p> +The following rules will do the trick: +</p> +<blockquote> +<tt> +Match expression = \+31([0-9]*) , Replace = 0$1<br> +Match expression = \+([0-9]*) , Replace = 00$1</br> +</tt> +</blockquote> +<h3>Example 2</h3> +<p> +You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. +</p> +<blockquote> +<tt> +Match expression = 0[0-9]* , Replace = 9$&<br> +</tt> +</blockquote></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 unencrypted.</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>Unsolicited</H3> +<p> +Asterisk provides unsolicited message waiting indication. +</p> +<H3>Solicited</H3> +<p> +Solicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsolicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For solicited 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> </context> <context> <name>WizardForm</name> @@ -5724,7 +5845,7 @@ Tento údaj je povinný.</translation> </message> <message> <source>SIP pre&xy:</source> - <translation>SIP pro&xy:</translation> + <translation type="vanished">SIP pro&xy:</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> @@ -5792,6 +5913,10 @@ V prípade priameho volania medzi IP adresami se môže jednať o meno hostiteľ <source>Invalid value for STUN server.</source> <translation>Neplatná hodnota pre STUN server.</translation> </message> + <message> + <source>SIP pro&xy:</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>YesNoDialog</name> diff --git a/src/gui/lang/twinkle_sv.ts b/src/gui/lang/twinkle_sv.ts index d8b0df6..9d7fd83 100644 --- a/src/gui/lang/twinkle_sv.ts +++ b/src/gui/lang/twinkle_sv.ts @@ -1,6 +1,6 @@ <?xml version="1.0" encoding="utf-8"?> <!DOCTYPE TS> -<TS version="2.0"> +<TS version="2.1" language="sv" sourcelanguage="en"> <context> <name>AddressCardForm</name> <message> @@ -962,6 +962,14 @@ <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> + <message> + <source>Are you sure you want to delete contact '%1' from the local address book?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete contact</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>GetProfileNameForm</name> @@ -4080,11 +4088,11 @@ och skapa följande statiska bindningar (UDP) i din NAT.</translation> </message> <message> <source>&Domain*:</source> - <translation>&Domän*:</translation> + <translation type="vanished">&Domän*:</translation> </message> <message> <source>Or&ganization:</source> - <translation>Or&ganisation:</translation> + <translation type="vanished">Or&ganisation:</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. @@ -4155,10 +4163,6 @@ Detta fält är obligatoriskt.</translation> <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> @@ -4583,10 +4587,6 @@ Send DTMF out-of-band via a SIP INFO request. <translation>Protokollalternativ</translation> </message> <message> - <source>Call &Hold variant:</source> - <translation type="unfinished"></translation> - </message> - <message> <source>RFC 2543</source> <translation>RFC 2543</translation> </message> @@ -4713,10 +4713,6 @@ This format is what most SIP phones use. <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> @@ -4760,10 +4756,6 @@ This format is what most SIP phones use. <translation>Ersätter</translation> </message> <message> - <source>Indicates if the Replaces-extenstion is supported.</source> - <translation type="unfinished"></translation> - </message> - <message> <source>REFER</source> <translation>REFER</translation> </message> @@ -4828,10 +4820,6 @@ This format is what most SIP phones use. <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>Integritet</translation> </message> @@ -4874,7 +4862,7 @@ When you choose this option you have to create static address mappings in your N </message> <message> <source>S&TUN server:</source> - <translation>S&TUN-server:</translation> + <translation type="vanished">S&TUN-server:</translation> </message> <message> <source>The hostname, domain name or IP address of the STUN server.</source> @@ -4937,40 +4925,6 @@ When you choose this option you have to create static address mappings in your N <translation>Ersätt</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> @@ -5232,10 +5186,6 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v <translation>&Aktivera ZRTP/SRTP-kryptering</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>Inställningar för ZRTP</translation> </message> @@ -5272,37 +5222,10 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v <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> @@ -5315,10 +5238,6 @@ Sollicited message waiting indication as specified by RFC 3842. <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> @@ -5511,10 +5430,6 @@ Detta kan vara värdnamnet eller IP-adressen för din dator, om du vill ha direk <translation type="unfinished"></translation> </message> <message> - <source>Use &STUN (does not work for incoming TCP)</source> - <translation type="unfinished"></translation> - </message> - <message> <source>P&ersistent TCP connection</source> <translation type="unfinished"></translation> </message> @@ -5690,6 +5605,119 @@ Detta kan vara värdnamnet eller IP-adressen för din dator, om du vill ha direk <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> + <message> + <source>Do&main*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Organi&zation:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>E&xpiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call Hold &variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max redirections:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extension is supported.</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 endpoint.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Asserted-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use STUN (does not wor&k for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>STUN ser&ver:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +</p> +<p> +For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. +</p> +<p> +The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. +</p> +<h3>Example 1</h3> +<p> +Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. +</p> +<p> +The following rules will do the trick: +</p> +<blockquote> +<tt> +Match expression = \+31([0-9]*) , Replace = 0$1<br> +Match expression = \+([0-9]*) , Replace = 00$1</br> +</tt> +</blockquote> +<h3>Example 2</h3> +<p> +You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. +</p> +<blockquote> +<tt> +Match expression = 0[0-9]* , Replace = 9$&<br> +</tt> +</blockquote></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 unencrypted.</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>Unsolicited</H3> +<p> +Asterisk provides unsolicited message waiting indication. +</p> +<H3>Solicited</H3> +<p> +Solicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsolicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For solicited 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> </context> <context> <name>WizardForm</name> diff --git a/src/gui/lang/twinkle_xx.ts b/src/gui/lang/twinkle_xx.ts index 067913e..1b92c88 100644 --- a/src/gui/lang/twinkle_xx.ts +++ b/src/gui/lang/twinkle_xx.ts @@ -1,6 +1,6 @@ <?xml version="1.0" encoding="utf-8"?> <!DOCTYPE TS> -<TS version="2.0"> +<TS version="2.1"> <context> <name>AddressCardForm</name> <message> @@ -919,6 +919,14 @@ <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> + <message> + <source>Are you sure you want to delete contact '%1' from the local address book?</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Delete contact</source> + <translation type="unfinished"></translation> + </message> </context> <context> <name>GetProfileNameForm</name> @@ -3855,14 +3863,6 @@ and create the following static bindings (UDP) in your NAT.</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> @@ -3927,10 +3927,6 @@ This field is mandatory.</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> @@ -4351,10 +4347,6 @@ Send DTMF out-of-band via a SIP INFO request. <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> @@ -4481,10 +4473,6 @@ This format is what most SIP phones use. <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> @@ -4528,10 +4516,6 @@ This format is what most SIP phones use. <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> @@ -4596,10 +4580,6 @@ This format is what most SIP phones use. <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> @@ -4665,18 +4645,10 @@ When you choose this option you have to create static address mappings in your N <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> @@ -4737,40 +4709,6 @@ When you choose this option you have to create static address mappings in your N <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> @@ -5040,10 +4978,6 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v <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> @@ -5080,37 +5014,10 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v <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> @@ -5123,10 +5030,6 @@ Sollicited message waiting indication as specified by RFC 3842. <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> @@ -5421,6 +5324,119 @@ This could be the hostname or IP address of your PC if you want direct PC to PC <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> + <message> + <source>Do&main*:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Organi&zation:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>E&xpiry:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Call Hold &variant:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Max redirections:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Indicates if the Replaces-extension is supported.</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 endpoint.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding.</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>&Send P-Asserted-Identity header when hiding user identity</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Use STUN (does not wor&k for incoming TCP)</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>STUN ser&ver:</source> + <translation type="unfinished"></translation> + </message> + <message> + <source><p> +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +</p> +<p> +For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. +</p> +<p> +The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. +</p> +<h3>Example 1</h3> +<p> +Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. +</p> +<p> +The following rules will do the trick: +</p> +<blockquote> +<tt> +Match expression = \+31([0-9]*) , Replace = 0$1<br> +Match expression = \+([0-9]*) , Replace = 00$1</br> +</tt> +</blockquote> +<h3>Example 2</h3> +<p> +You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. +</p> +<blockquote> +<tt> +Match expression = 0[0-9]* , Replace = 9$&<br> +</tt> +</blockquote></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 unencrypted.</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>Unsolicited</H3> +<p> +Asterisk provides unsolicited message waiting indication. +</p> +<H3>Solicited</H3> +<p> +Solicited message waiting indication as specified by RFC 3842. +</p></source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Unsolicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>Solicited MWI</source> + <translation type="unfinished"></translation> + </message> + <message> + <source>For solicited 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> </context> <context> <name>WizardForm</name> |