Skip to content
Snippets Groups Projects
Select Git revision
  • fix_fallback_from_tls
  • asterisk_rdkb
  • fix_fallback
  • 14666_fxs_no_wideband_codec
  • fix_srv_records
  • wenpeng-test-250121
  • wenpeng-registration-test
  • fix_deadlock_in_bridge_peer_functions
  • devel default protected
  • securesip_pjproject_part1
  • PJ_Example_Review
  • fix_re_register_authorization_header
  • fix_internal_calls_auth
  • fix_internal_calls
  • Secure_SIPtelephony_part1
  • SIPtelephony_DECT_syslog
  • syslog_develop_phase1
  • merge-asterisk-20.8.1
  • asterisk_rdkb_temp
  • bug_14682_voicemail
  • 22.0.0-pre1
  • 21.4.2
  • 20.9.2
  • 18.24.2
  • certified-20.7-cert2
  • certified-18.9-cert11
  • 21.4.1
  • 20.9.1
  • 18.24.1
  • 21.4.0
  • 20.9.0
  • 18.24.0
  • certified-20.7-cert1
  • certified-18.9-cert10
  • 21.4.0-rc1
  • 20.9.0-rc1
  • 18.24.0-rc1
  • 21.3.1
  • 20.8.1
  • 18.23.1
40 results
Search by author
  • Any Author
  • Adam Porteous Adam Porteous adam.porteous
  • Alex Hopkins Alex Hopkins 59hopper
  • Amin Ben Romdhane Amin Ben Romdhane amin.benromdhane
  • Andreas Gnau Andreas Gnau AGnau
  • Andrej Kucharik Andrej Kucharik andrej.kucharik
  • Andrii Pypa Andrii Pypa andrii.pypa
  • Anjan Chanda Anjan Chanda anjancx
  • Balalakshmi Arunachalam Rajendran Balalakshmi Arunachalam Rajendran arbala
  • Benjamin Larsson Benjamin Larsson benjamin
  • Charles Foster Charles Foster charles.foster_bt.com
  • Dariusz Iwanoczko Dariusz Iwanoczko dariusz.iwanoczko
  • Divya Chandrasekaran Divya Chandrasekaran divya.c
  • Dominic Lake Dominic Lake dominic.lake
  • Elena Vengerova Elena Vengerova helen
  • Erik Karlsson Erik Karlsson erik.karlsson
  • Filip Matusiak Filip Matusiak filmat
  • Fredrik Östling Fredrik Östling fredrik.ostling
  • Geoff Elliott Geoff Elliott gdelliott
  • George Yang George Yang gyang
  • Gouri Natarajan Gouri Natarajan gouri.natarajan
20 authors
  1. May 09, 2014
  3. Feb 05, 2014
  5. Aug 08, 2013
    • Matthew Jordan's avatar
      Improve disk writes for wav49 format · 16fd65bb
      Matthew Jordan authored 
      Writing to a file in the wav49 format performs rather inefficiently. The
procedure is approximately:
 (1) Write GSM frame to the end of the file
 (2) Seek to the end of the file
 (3) Seek to the header
 (4) Update the file size
 (5) Seek (again) to the end of the file
 (6) Repeat

This pattern negates any attempt to use the stdio buffering setup in
ast_writefile. It also results in many small writes that require a seek going
to the disk each second which translates to poor disk performance on certain
file systems, particularly when there are multiple wav49 files being written
simultaneously.

(closes issue ASTERISK-19595)
Reported by: Byron Clark
Tested by: Byron Clark
patches:
  gsm_wav_only_update_header_on_close.patch uploaded by byronclark (License 6157)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      16fd65bb
  7. Oct 14, 2012
  9. Aug 30, 2012
    • Matthew Jordan's avatar
      Clean up doxygen warnings · 8018b879
      Matthew Jordan authored 
      This patch fixes numerous doxygen warnings across Asterisk.  It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.

Much thanks to Andrew for tackling one of the Asterisk janitor projects!

(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
  doxygen_partial.diff uploaded by Andrew Latham (license 5985)
  make_progdocs.diff uploaded by Andrew Latham (license 5985)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8018b879
  11. Jun 07, 2012
  13. Apr 29, 2012
  15. Apr 17, 2012
  17. Apr 16, 2012
  19. Apr 06, 2012
  21. Feb 16, 2012
  23. Feb 14, 2012
  25. Feb 08, 2012
  27. Nov 09, 2011
  29. Jul 29, 2011
  31. Jul 14, 2011
  33. Jul 08, 2011
  35. Jul 07, 2011
  37. May 16, 2011
  39. May 03, 2011
  41. Apr 25, 2011
  43. Feb 22, 2011
    • David Vossel's avatar
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd... · d760e81f
      David Vossel authored 
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d760e81f
  45. Feb 03, 2011
  47. Sep 02, 2010
  49. Jul 26, 2010
  51. Jul 20, 2010
  53. Jun 16, 2010
  55. Apr 20, 2010
  57. Dec 08, 2009
    • Russell Bryant's avatar
      Set a module load priority for format modules. · 8ab22f5d
      Russell Bryant authored 
      A recent change to app_voicemail made it such that the module now assumes that
all format modules are available while processing voicemail configuration.
However, when autoloading modules, it was possible that app_voicemail was
loaded before the format modules.  Since format modules don't depend on
anything, set a module load priority on them to ensure that they get loaded
first when autoloading.

This fix applies to trunk, 1.6.1, and 1.6.2.  The fix for 1.4 and 1.6.0 will
require a different approach since the module load priority functionality is
not present in the module API.

(issue #16412)
Reported by: jiddings


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8ab22f5d
  59. Nov 04, 2009
  61. Oct 19, 2009
  63. Jun 15, 2009
  65. May 21, 2009
    • Kevin P. Fleming's avatar
      Const-ify the world (or at least a good part of it) · e6b2e9a7
      Kevin P. Fleming authored 
      This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e6b2e9a7
  67. Apr 08, 2009
  69. Feb 15, 2009
  71. Feb 13, 2009
    • Kevin P. Fleming's avatar
      Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C... · 2a53f2ec
      Kevin P. Fleming authored 
      Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)

This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.

Along the way, some related work was done:

1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.

2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.

3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).

Review: http://reviewboard.digium.com/r/158/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2a53f2ec