asterisk-opus 13.7+20161113-4 source package in Ubuntu


asterisk-opus (13.7+20161113-4) unstable; urgency=medium

  * Modernize cdbs:
    + Do copyright-check in maintainer script (not during build).
      Stop build-depend on licensecheck.
  * Modernize Vcs-* fields: Use git (not cgit) in path.
  * Declare compliance with Debian Policy 4.1.0.
  * Update copyright info:
    + Use https protocol in file format URL.
    + Extend coverage for myself.
  * Advertise DEP-3 format in patch headers.
  * This (simply rebuilding) fixes linking with current Asterisk.
    Closes: Bug#872760. Thanks to Sam Hartman.

 -- Jonas Smedegaard <email address hidden>  Mon, 28 Aug 2017 10:32:42 +0200

Upload details

Uploaded by:
Debian VoIP Team on 2017-08-28
Uploaded to:
Original maintainer:
Debian VoIP Team
Medium Urgency

See full publishing history Publishing

Series Pocket Published Component Section
Bionic release on 2017-11-02 universe misc


File Size SHA-256 Checksum
asterisk-opus_13.7+20161113-4.dsc 2.0 KiB a8d2ab5d936bc2e895a6ba754833899686e4dcb28fbad7eb1292a0917b66d1e3
asterisk-opus_13.7+20161113.orig.tar.gz 23.3 KiB 869ccf8fc91aa5917cf69e434bab61547bd761dfcc55e889e432330ed4c2ba8c
asterisk-opus_13.7+20161113-4.debian.tar.xz 5.1 KiB 969553ed4130a329eb27290a5360c07be9428be6bf7237fbc7c3e2cd652e2472

No changes file available.

Binary packages built by this source

asterisk-opus: 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).

asterisk-opus-dbgsym: No summary available for asterisk-opus-dbgsym in ubuntu cosmic.

No description available for asterisk-opus-dbgsym in ubuntu cosmic.