Your message dated Wed, 21 Dec 2016 13:00:11 +0000
with message-id <e1cjgv9-000d5y...@fasolo.debian.org>
and subject line Bug#848560: fixed in asterisk-opus 13.7+20161113-1
has caused the Debian Bug report #848560,
regarding ITP: asterisk-opus -- opus module for Asterisk
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
message is talking about, this may indicate a serious mail system
misconfiguration somewhere. Please contact ow...@bugs.debian.org
immediately.)


-- 
848560: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=848560
Debian Bug Tracking System
Contact ow...@bugs.debian.org with problems
--- Begin Message ---
Package: wnpp
Severity: wishlist
Owner: Jonas Smedegaard <d...@jones.dk>

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* Package name    : asterisk-opus
  Version         : 13.7+20161113
  Upstream Author : Lorenzo Miniero
* URL             : https://github.com/traud/asterisk-opus
* License         : GPL-2
  Programming Lang: C
  Description     : opus module for Asterisk

 Module for the Asterisk open source PBX which allows you to use the
 Opus audio codec.
 .
 Opus is the default audio codec in WebRTC. WebRTC is available in
 Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
 for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
 codecs like CELT and SiLK. Furthermore in favor of Opus, other
 open-source audio codecs are no longer developed, like Speex, iSAC,
 iLBC, and Siren. If you use your Asterisk as a back-to-back user agent
 (B2BUA) and you transcode between various audio codecs, one should
 enable Opus for future compatibility.
 .
 Opus is not only supported for pass-through but can be transcoded as
 well. This allows you to translate to/from other audio codecs like
 those for landline telephones (ISDN: G.711; DECT: G.726-32; and HD:
 G.722) or mobile phones (GSM, AMR, AMR-WB, 3GPP EVS).

Opus module is currently included as a patch to Asterisk itself, but
that complicates security tracking (see bug#847666), and same codebase
has recently grown support for building out-of-tree as well.

This package will be maintained in the VoIP team.

 - Jonas

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--- Begin Message ---
Source: asterisk-opus
Source-Version: 13.7+20161113-1

We believe that the bug you reported is fixed in the latest version of
asterisk-opus, which is due to be installed in the Debian FTP archive.

A summary of the changes between this version and the previous one is
attached.

Thank you for reporting the bug, which will now be closed.  If you
have further comments please address them to 848...@bugs.debian.org,
and the maintainer will reopen the bug report if appropriate.

Debian distribution maintenance software
pp.
Jonas Smedegaard <d...@jones.dk> (supplier of updated asterisk-opus package)

(This message was generated automatically at their request; if you
believe that there is a problem with it please contact the archive
administrators by mailing ftpmas...@ftp-master.debian.org)


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Format: 1.8
Date: Sun, 18 Dec 2016 13:03:07 +0100
Source: asterisk-opus
Binary: asterisk-opus
Architecture: source amd64
Version: 13.7+20161113-1
Distribution: unstable
Urgency: low
Maintainer: Debian VoIP Team <pkg-voip-maintain...@lists.alioth.debian.org>
Changed-By: Jonas Smedegaard <d...@jones.dk>
Description:
 asterisk-opus - opus module for Asterisk
Closes: 848560
Changes:
 asterisk-opus (13.7+20161113-1) unstable; urgency=low
 .
   * Initial release.
     Closes: bug#848560.
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