---
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'authentification par hachage
MD5 pour toutes les requêtes SIP Prise en charge d'authentification par hachage AKAv1-MD5 pour toutes les requêtes
SIP (nouveau) Masquage d'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'autre
bout de la ligne. Contrôle de gain automatique (AGC) (nouveau) Réduction de bruit (nouveau) Détection d'activité
vocale (VAD) (nouveau) Contrôle d'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
