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Bug#738469: marked as done (RFA: twinkle -- Voice over Internet Protocol (VoIP) SIP Phone)

Your message dated Tue, 5 Jan 2016 19:33:26 +0800
with message-id <568BAA06.7090900@rolf.leggewie.biz>
and subject line Re: RFA: twinkle -- Voice over Internet Protocol (VoIP) SIP Phone
has caused the Debian Bug report #738469,
regarding RFA: twinkle -- Voice over Internet Protocol (VoIP) SIP Phone
to be marked as done.

This means that you claim that the problem has been dealt with.
If this is not the case it is now your responsibility to reopen the
Bug report if necessary, and/or fix the problem forthwith.

(NB: If you are a system administrator and have no idea what this
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immediately.)


-- 
738469: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=738469
Debian Bug Tracking System
Contact owner@bugs.debian.org with problems
--- Begin Message --- 
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-maintainers@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)

--- End Message ---
--- Begin Message --- 
On 30.12.2015 07:19, Peter Colberg wrote:
> Dear Debian VoIP team,
>
> Twinkle, a graphical SIP client, has recently been resurrected upstream
> with a port to Qt 5 [1]. I have packaged twinkle 1.9.0 for Debian [2],
> which has been uploaded by the previous maintainer, Clint Adams.
>
> [1] http://twinkle.dolezel.info/
> [2] https://bugs.debian.org/797159

Thanks, Peter, that is great news.

I'd be willing to join your efforts to maintain twinkle in Debian.  I am
DM and thus would eventually be able to make the uploads.  Just put my
name in the uploaders field in your next upload if you want. You might
think about adding the VoIP team there as well.

The RFA #738469 has been dealt with by your upload of so I will close
that ticket.

- Rolf

--- End Message ---
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