Package: wnpp Severity: normal I request an adopter for the twinkle package.
The package description is: Soft-phone for making telephone calls using SIP over an IP network. . Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls. . In addition to making basic voice calls Twinkle provides you the following features regardless of the services that your VoIP service provider might offer. . 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) . 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 . 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) . 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) Description-md5: e13a5ad5bec99b1338d6a8de9eb67858 Homepage: http://www.twinklephone.com/ Section: comm Priority: optional Filename: pool/main/t/twinkle/twinkle_1.4.2-4_amd64.deb Size: 698384 MD5sum: 745ec08fb8f3215d7d011785516eaf9f SHA1: 6974cd31af049a7d9ed2b7dbce2c2bc6e696a90e SHA256: 2f0e684885ea2b80936a768bf16445d25fe21e7aec9b77a568eb8e5152df547d Package: twinkle Priority: optional Section: comm Installed-Size: 4640 Maintainer: Debian VoIP Team <pkg-voip-maintain...@lists.alioth.debian.org> Architecture: amd64 Source: twinkle (1:1.4.2-2) Version: 1:1.4.2-2+b2 Depends: libasound2 (>> 1.0.18), libboost-regex1.42.0 (>= 1.42.0-1), libc6 (>= 2.9), libccgnu2-1.7-0, libccrtp1-1.7-0, libgcc1 (>= 1:4.1.1), libgsm1 (>= 1.0.13), libmagic1, libncurses5 (>= 5.7+20100313), libqt3-mt (>= 3:3.3.8b), libreadline6 (>= 6.0), libsndfile1 (>= 1.0.20), libspeex1 (>= 1.2~beta3-1), libspeexdsp1 (>= 1.2~beta3.2-1), libstdc++6 (>= 4.4.0), libx11-6 (>= 0), libxext6, libxml2 (>= 2.7.4), libzrtpcpp-1.4-0, zlib1g (>= 1:1.1.4) Filename: pool/main/t/twinkle/twinkle_1.4.2-2+b2_amd64.deb Size: 1758644 MD5sum: 8537211787283bd9dcac949b90f215a7 SHA1: d1bdb30da29e44a2e57057e8a11b303745d37854 SHA256: d480881ae14fca567dcc8a2df9a34935b46b5fc4a8222ccb73db2d31e0ede7b9 Description-en: Voice over Internet Protocol (VoIP) SIP Phone Soft-phone for making telephone calls using SIP over an IP network. . Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls. . In addition to making basic voice calls Twinkle provides you the following features regardless of the services that your VoIP service provider might offer. . 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) . 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 . 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) . 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) -- To UNSUBSCRIBE, email to debian-wnpp-requ...@lists.debian.org with a subject of "unsubscribe". Trouble? Contact listmas...@lists.debian.org Archive: http://lists.debian.org/20140209211828.ga7...@scru.org
