⇦ | twinkle [universe]
Last updated on: 2018-04-27 01:33 [UTC]

Metadata for twinkle in universe

twinkle.desktop - 1:1.10.1+dfsg-3 ⚙ amd64 ⚙ armhf ⚙ arm64 ⚙ i386 ⚙ ppc64el ⚙ s390x

Icon
---
Type: desktop-application
ID: twinkle.desktop
Package: twinkle
Name:
  C: Twinkle
Summary:
  de: VoIP-SIP-Softwaretelefon
  pl: Aplikacja telefoniczna do VoIP (Voice over Internet Protocol), wykorzystująca SIP
  pt_BR: Telefone SIP de voz sobre protocolo de internet (VoIP)
  ja: Voice over Internet Protocol (VoIP) SIP 電話
  C: Voice over Internet Protocol (VoIP) SIP Phone
  uk: VoIP (передача голосу по IP-протоколу) SIP-телефон
  pt: Telefone SIP Voz sobre IP (Voice over Internet Protocol - VoIP)
  cs: VoIP (Voice over Internet Protocol) telefon SIP
  ru: VoIP (передача голоса по IP-протоколу) SIP-телефон
  ko: Voice over Internet Protocol (VoIP) SIP 폰
  it: Telefono SIP VoIP (Voice over Internet Protocol)
  da: Voice over Internet Protocol (VoIP) SIP-telefon
Description:
  de: >-
    <p>Software-Telefon, um mittels SIP Telefonate über ein IP-Netzwerk zu führen.</p>

    <p>Twinkle unterstützt direkte Kommunikation von IP-Telefon zu IP-Telefon oder über ein Netzwerk, das einen SIP-Proxy
    zur Weiterleitung der Anrufe nutzt.</p>

    <p>Zusätzlich zum einfachen Telefonieren bietet Twinkle die folgenden Funktionen, unabhängig davon, ob Ihr VoIP-Dienstleister
    Ihnen diese Dienstleistungen anbietet.</p>

    <p> 2 eingehende Anrufkanäle (Leitungen)  mehrere aktive Telefonidentitäten  benutzerdefinierte Rufzeichen  Anklopfen 
    Verbindung halten  3-Wege-Konferenzschaltung  Stummfunktion  Rufumleitung bei Anforderung  bedingungslose Rufumleitung 
    Rufumleitung wenn besetzt  Rufumleitung wenn niemand abnimmt  Rufumleitungsanfrage ablehnen  blinde Rufweiterleitung 
    Rufweiterleitung nach Gesprächsannahme (betreute Rufweiterleitung) (neu)  Rufweiterleitungsanfrage ablehnen  Anruf ablehnen 
    Wahlwiederholung  »Nicht stören«-Funktion  automatische Rufannahme  Benachrichtigung über Nachrichten in Abwesenheit 
    Zielwahl für Sprachnachrichten  benutzerdefinierte Skripte bei Anrufereignissen  z.B. zur zielgerichteten Rufablehnung
    oder für charakteristische Klingeltöne  RFC 2833 DTMF-Ereignisse  In-band-DTMF  Out-of-band-DTMF (SIP INFO)  STUN-Unterstützung
    für Netzwerke mit NAT  sende NAT-Pakete zur Verbindungshaltung mit STUN  NAT-Unterstützung für statische Netze  Meldung
    verlorener Anrufe  Geschichte der Kommunikationsdatensätze für eingehende, ausgehende, erfolgreiche und fehlgeschlagenen
    Verbindungen  Unterstützung für DNS SRV  automatischer Serverwechsel, wenn ein Server nicht verfügbar ist  andere Anwendungen
    können einen Anruf mit Twinkle aufbauen, z.B. Anruf aus einem Adressbuch  Bild im Systembenachrichtigungsabschnitt  Systembenachrichtigungsmenü
    zum Aufbau und Annahme eines Anrufs mit ausgeblendetem Twinkle  benutzerdefinierte Anzahl an Konvertierungsvorschriften 
    einfaches Adressbuch  Unterstützung für UDP und TCP (neu) zur Übertragung von SIP  Präsenz  Sofortnachrichtendienst  einfacher
    Dateiaustausch mittels des Sofortnachrichtendienst  Benachrichtigung über das Verfassen einer Sofortnachricht des Gesprächspartners 
    Befehlszeilenschnittstelle (CLI)</p>

    <p>VoIP-Sicherheit  sichere Sprachübetragung mit ZRTP/SRTP  Unterstützung von MD5-Hash-Authentifizierung für alle SIP-Anfragen 
    Unterstützung von AKAv1-MD5-Hash-Authentifizierung für alle SIP-Anfragen (neu)  Rufnummernunterdrückung</p>

    <p>Audio-Codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>Für alle Codecs sind folgende Vorverarbeitungsoptionen verfügbar, um die Qualität am anderen Ende der Leitung zu verbessern. 
    Lautstärkeanpassung (AGC) (neu)  Rauschreduzierung (neu)  Erkennung der Sprachaktivität (VAD) (neu)  Echoüberwachung (AEC)
    [experimentell] (neu)</p>
  pt_BR: >-
    <p>Telefone virtual para fazer ligações telefônicas usando SIP sobre rede IP.</p>

    <p>Twinkle tem suporte a comunicação direta de telefone IP para telefone IP ou uma rede usando proxy SIP para rotear chamadas.</p>

    <p>Além de fazer chamadas de voz básicas, Twinkle fornece  para você os seguintes recursos além dos serviços que seu provedor
    de VoIP pode oferecer.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  sl: >-
    <p>Programski telefon za telefonske klice s SIP preko omrežja IP.</p>

    <p>Twinkle podpira neposredno sporazumevanje od telefona IP do telefona IP ali omrežja z uporabo posredniškega strežnika
    SIP za usmeritev vaših klicev.</p>

    <p>Poleg osnovnih zvokovnih klicev vam Twinkle zagotavlja naslednje zmožnosti ne glede na storitve, ki jih morda ponuja
    ponudnik storitve VoIP.</p>

    <p> 2 videza klicev (liniji)  več določil dejavnih klicev  melodije zvonjenja po meri  čakajoči klic  zadržani klic  3
    smerno usmerjanje konference  nemo  preusmeritev klicev na zahtevo  brezpogojna preusmeritev klicev  preusmeritev klicev
    ob zaposlenosti  preusmeritev klicev, če ni odgovora  zahteva preusmeritve blokiranega klica  prenos slepih klicev  ponovi
    zadnji klic  ne moti  samodejen odgovor  pokazatelj čakajočega sporočila  hitra številčnica glasovne pošte  uporabniško
    določljivi skripti, ki se sprožijo ob klicih   na primer podpora selektivnega zvonenja  dogodki RFC 2833 DMTF  DMTF v
    pasu  DMTF iz pasu (SIP INFO)  podpora STUN za prehod NAT  pošiljanje paketov NAT keep alive pri uporabi STUN  prehod
    NAT skozi statično določne  pokazatelj zgrešenih klicev  zgodovina podrobnosti klicev za prihajajoče, odhajajoče, uspešne
    in zgrešene klice  podpora DNS SRV  samodejno preklapljanje za nadomestni strežnik, če strežnik ni dosegljiv  drugi programi
    lahko kličejo preko Twinkle, na primer kličejo preko imenika  ikona sistemske vrstice  meni sistemske vrstice za izvir
    in sprejem klicev, ko ostane Twinkle skrit  uporabniško določljiva pravila pretvorbe  enostaven imenik  podpora za UDP
    in TCP (nov) kot prenos za SIP  prisotnost  hipno sporočanje  enostaven prenos datotek preko hipnega sporočanja  pokazatelj
    sestavljanja hipnega sporočila  vmesnik ukazne vrstice (CLI)</p>

    <p>Varnost VoIP  Varno sporazumevanje z ZRTP/SRTP  Podpora izvlečkov overitve MD5 za vse zahteve SIP  Podpora izvlečkov
    overitce AKAv1-MD5 za vse zahteve SIP (novo)  Skrivanje identitete</p>

    <p>Zvočni kodeki  G.711 A-law (64 kbps obremenitev, 8 kHz hitrost vzorčenja)  G.711 u-law (64 kbps obremenitev, 8 kHz
    hitrost vzorčenja)  GSM (13 kbps obremenitev, 8 kHz hitrost vzorčenja)  Speex narrow band (15.2 kbps obremenitev, 8 kHz
    hitrost vzorčenja)  Speex wide band (28 kbps obremenitev, 16 kHz hitrost vzorčenja)  Speex ultra wide band (36 kbps obremenitev,
    32 kHz hitrost vzorčenja)  G.726 (16, 24, 32 or 40 kbps obremenitev, 8 kHz hitrost vzorčenja)</p>

    <p>Za vse kodeke so na voljo naslednje možnosti predobdelovanja za izboljšanje kakovosti daljnega klica.  samodejen nadzor
    glasnosti (AGC) (novo)  zmanjšanje šuma (novo)  zaznavanje dejavnosti glasu (VAD) (novo)  akustični nadzor odmevov (AEC)
    [preizkusno] (novo)</p>
  pt: >-
    <p>Soft-phone para fazer chamadas telefónicas usando SIP sobre uma rede IP.</p>

    <p>O Twinkle suporta comunicação direta telefone IP a telefone IP ou numa rede usando um proxy SIP para encaminhar as
    suas chamadas.</p>

    <p>Além de fazer chamadas de voz básicas, o Twinkle pode oferecer as seguintes funcionalidades, independente dos serviços
    que o seu provedor possa oferecer.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  uk: >-
    <p>Програмний телефон для здійснення дзвінків через IP-мережу.</p>

    <p>Twinkle підтримує як пряме підключення з IP-телефону на IP-телефон, так і з’єднання через мережу, використовуючи SIP-проксі
    для спрямування дзвінка.</p>

    <p>Для здійснення простих голосових дзвінків Twinkle надає наступні можливості, навіть якщо провайдер VoIP цього не підтримує.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  C: >-
    <p>Twinkle is a soft-phone for making telephone calls over an IP network using the SIP protocol. You can use it for direct
    IP phone to IP phone communication or in a network using a SIP proxy to route your calls. Notable features include multiple
    active identities, call transfer, call rejection, 2 simultaneous calls and 3-way conference calls.</p>

    <p>This package contains the graphical interface.</p>
  en: >-
    <p>Twinkle is a soft-phone for making telephone calls over an IP network using the SIP protocol. You can use it for direct
    IP phone to IP phone communication or in a network using a SIP proxy to route your calls. Notable features include multiple
    active identities, call transfer, call rejection, 2 simultaneous calls and 3-way conference calls.</p>

    <p>This package contains the graphical interface.</p>
  en_CA: >-
    <p>Soft-phone for making telephone calls using SIP over an IP network.</p>

    <p>Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls.</p>

    <p>In addition to making basic voice calls Twinkle provides you the following features regardless of the services that
    your VoIP service provider might offer.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  ru: >-
    <p>Программный телефон для осуществления звонков, через IP-сеть.</p>

    <p>Twinkle поддерживает как прямое подключение с IP-телефона на IP-телефон, так и соединение по сети, используя SIP-прокси
    для направления звонка.</p>

    <p>Для осуществления простых голосовых звонков Twinkle предоставляет следующие возможности, даже если провайдер VoIP этого
    не поддерживает.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  fr: >-
    <p>Soft-phone pour passer des appels téléphoniques en utilisant SIP sur un réseau IP.</p>

    <p>Twinkle prend en charge la communication directe de téléphone IP à téléphone IP ou un réseau utilisant un proxy SIP
    pour acheminer vos appels.</p>

    <p>En plus des appels vocaux de base, Twinkle vous offre les fonctionnalités suivantes quels que soient les services que
    votre fournisseur de service VoIP peut offrir.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>Sécurité VoIP  Sécurise la communication vocale par ZRTP/SRTP  Prise en charge d&apos;authentification par hachage
    MD5 pour toutes les requêtes SIP  Prise en charge d&apos;authentification par hachage AKAv1-MD5 pour toutes les requêtes
    SIP (nouveau)  Masquage d&apos;identité</p>

    <p>Codecs audio  G.711 A-law (64 kbps de charge utile, 8 kHz de fréquence d’échantillonnage)  G.711 u-law (64 kbps de
    charge utile, 8 kHz de fréquence d’échantillonnage)  GSM (13 kbps de charge utile, 8 kHz de fréquence d’échantillonnage) 
    Speex bande étroite (15.2 kbps de charge utile, 8 kHz de fréquence d’échantillonnage)  Speex bande large (28 kbps de charge
    utile, 16 kHz de fréquence d’échantillonnage)  Speex bande ultra large (36 kbps de charge utile, 32 kHz de fréquence d’échantillonnage) 
    G.726 (16, 24, 32 ou 40 kbps de charge utile, 8 kHz de fréquence d’échantillonnage)</p>

    <p>Pour tous les codecs, les options de prétraitement suivantes sont disponibles pour améliorer la qualité à l&apos;autre
    bout de la ligne.  Contrôle de gain automatique (AGC) (nouveau)  Réduction de bruit (nouveau)  Détection d&apos;activité
    vocale (VAD) (nouveau)  Contrôle d&apos;écho acoustique (AEC) [expérimental] (nouveau)</p>
  ja: >-
    <p>IP ネットワーク上で SIP を用いて電話をかけるためのソフトウェア電話です。</p>

    <p>Twinkle は IP 電話から IP 電話への直接通話や、あなたのコールを送るめに SIP プロキシを使うネットワークに対応しています。</p>

    <p>基本的な音声コールの作成に加え、Twinkle は VoIP サービスプロバイダが提 供するサービスに関係なく、以下の機能を提供します。</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  en_GB: >-
    <p>Soft-phone for making telephone calls using SIP over an IP network.</p>

    <p>Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls.</p>

    <p>In addition to making basic voice calls Twinkle provides you the following features regardless of the services that
    your VoIP service provider might offer.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  it: >-
    <p>Telefono software per fare telefonate usando SIP su IP in rete.</p>

    <p>Twinkle supporta comunicazioni dirette da telefono IP a telefono IP o ad una rete usando un proxy SIP per instradare
    le chiamate.</p>

    <p>In aggiunta alle chiamate voce base, Twinkle fornisce le seguenti funzionalità indipendentemente dagli eventuali servizi
    del proprio provider VoIP.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
  en_AU: >-
    <p>Soft-phone for making telephone calls using SIP over an IP network.</p>

    <p>Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls.</p>

    <p>In addition to making basic voice calls Twinkle provides you the following features regardless of the services that
    your VoIP service provider might offer.</p>

    <p> 2 call appearances (lines)  Multiple active call identities  Custom ring tones  Call Waiting  Call Hold  3-way conference
    calling  Mute  Call redirection on demand  Call redirection unconditional  Call redirection when busy  Call redirection
    no answer  Reject call redirection request  Blind call transfer  Call transfer with consultation (attended call transfer)
    (new)  Reject call transfer request  Call reject  Repeat last call  Do not disturb  Auto answer  Message Waiting Indication 
    Voice mail speed dial  User definable scripts triggered on call events   E.g. to implement selective call reject or distinctive
    ringing  RFC 2833 DTMF events  In-band DTMF  Out-of-band DTMF (SIP INFO)  STUN support for NAT traversal  Send NAT keep
    alive packets when using STUN  NAT traversal through static provisioning  Missed call indication  History of call detail
    records for incoming, outgoing, successful and missed  DNS SRV support  Automatic fail-over to an alternate server if
    a server is unavailable  Other programs can originate a call via Twinkle, e.g. call from address book  System tray icon 
    System tray menu to originate and answer calls while Twinkle stays hidden  User definable number conversion rules  Simple
    address book  Support for UDP and TCP (new) as transport for SIP  Presence  Instant messaging  Simple file transfer with
    instant message  Instant message composition indication  Command line interface (CLI)</p>

    <p>VoIP security  Secure voice communication by ZRTP/SRTP  MD5 digest authentication support for all SIP requests  AKAv1-MD5
    digest authentication support for all SIP requests (new)  Identity hiding</p>

    <p>Audio codecs  G.711 A-law (64 kbps payload, 8 kHz sampling rate)  G.711 u-law (64 kbps payload, 8 kHz sampling rate) 
    GSM (13 kbps payload, 8 kHz sampling rate)  Speex narrow band (15.2 kbps payload, 8 kHz sampling rate)  Speex wide band
    (28 kbps payload, 16 kHz sampling rate)  Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)  G.726 (16, 24,
    32 or 40 kbps payload, 8 kHz sampling rate)</p>

    <p>For all codecs the following preprocessing options are available to improve quality at the far end of a call.  Automatic
    gain control (AGC) (new)  Noise reduction (new)  Voice activity detection (VAD) (new)  Acoustic echo control (AEC) [experimental]
    (new)</p>
Categories:
- Network
- Telephony
Icon:
  cached:
  - name: twinkle_twinkle.png
    width: 64
    height: 64
  - name: twinkle_twinkle.png
    width: 128
    height: 128