Skip to content
Snippets Groups Projects
Select Git revision
  • dev_fxs_no_wb
  • fix_transport_fallback_at_dns_failure
  • 16302-renew-registration
  • asterisk_rdkb
  • 14600-wenpeng
  • 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
  • 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
  • Filip Strelec Filip Strelec filip.strelec
  • Fredrik Östling Fredrik Östling fredrik.ostling
  • Geoff Elliott Geoff Elliott gdelliott
  • George Yang George Yang gyang
20 authors
  1. Apr 01, 2011
  3. Mar 28, 2011
  5. Mar 23, 2011
  7. Mar 22, 2011
  9. Mar 18, 2011
    • Jonathan Rose's avatar
      Adds an option to FollowMe that isn't useful for the bug it was made to solve.... · 18a6c3a4
      Jonathan Rose authored 
      Adds an option to FollowMe that isn't useful for the bug it was made to solve.  Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      18a6c3a4
    • Richard Mudgett's avatar
      Merged revisions 311295 via svnmerge from · 4a8c7797
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines
  
  Merged revision 310986 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
  
  ..........
    r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines
  
    Dial() o option broke when connected line feature added.
  
    The patch restores the o option behavior and adds the ability to specify
    the CallerID.  The Dial o and f options are complementary to each other.
    The o option stores the CallerID on the outgoing channel as the channel's
    CallerID.  The f option forces the CallerID sent by the outgoing channel.
  
    o(x) - The argument 'x' is optional.  If not present, then specify that
    the CallerID that was present on the *calling* channel be stored as the
    CallerID on the *called* channel.  This was the behavior of Asterisk 1.0
    and earlier.  If present, then specify the CallerID stored on the *called*
    channel.  Note that o(${CALLERID(all)}) is similar to option o without
    parameters.
  
    f(x) - The argument 'x' is optional and its presence changes the behavior
    of this option.  If not present, then force the outgoing CallerID on a
    call-forward or deflection to the dialplan extension for this Dial() using
    a dialplan 'hint'.  For example, some PSTNs do not allow CallerID to be
    set to anything other than the numbers assigned to you.  If present, then
    force the outgoing CallerID to 'x'.
  
    Patches:
  	jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
    Tested by: rmudgett
  
    JIRA ABE-2752
    JIRA SWP-3096
  ..........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4a8c7797
  11. Mar 17, 2011
  13. Mar 11, 2011
  15. Mar 10, 2011
  17. Mar 07, 2011
  19. Mar 04, 2011
  21. 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
  23. Feb 15, 2011
  25. Feb 14, 2011
    • Tilghman Lesher's avatar
      Merged revisions 307750 via svnmerge from · 7800a1c3
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines
  
  Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
  
  A bug in AEL did not distinguish between the "s" extension generated by
  AEL and an "s" extension that was required to exist by the chan_dahdi
  (or another channel) that was not supplied with a starting extension.
  Therefore, AEL made incorrect assumptions about what commands were
  permissable in the context.  This was fixed by making AEL generate a
  different extension name.  However, Dial and Queue make additional
  assumptions about the name of the default gosub extension.  Therefore,
  they needed to be brought into line with a "macro" rendered by AEL (as
  a gosub), without breaking traditional dialplans written without the
  aid of AEL.
  
  Related to (issue #18480)
   Reported by: nivek
  
  (closes issue #18729)
   Reported by: kkm
   Patches: 
         20110209__issue18729.diff.txt uploaded by tilghman (license 14)
         018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
   Tested by: kkm
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7800a1c3
  27. Feb 09, 2011
  29. Feb 08, 2011
  31. Feb 04, 2011
  33. Feb 03, 2011
  35. Feb 02, 2011
  37. Feb 01, 2011
  39. Jan 31, 2011
  41. Jan 30, 2011
  43. Jan 29, 2011
  45. Jan 28, 2011
  47. Jan 26, 2011
  49. Jan 25, 2011
    • Jeff Peeler's avatar
      Merged revisions 303678 via svnmerge from · d3c7a689
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines
  
  Merged revisions 303677 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
    
    Merged revisions 303676 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines
      
      Fix voicemail sequencing for file based storage.
      
      A previous change was made to account for when the number of voicemail messages
      exceeds the max limit to be handled properly, but it caused gaps in the messages
      to not be properly handled. This has now been resolved.
      
      In later non 1.4 branches, it appears that resequencing wasn't even occurring
      due from what appears and accidental code removal.
      
      (closes issue #18498)
      Reported by: JJCinAZ
      Patches: 
            bug18498v2.patch uploaded by jpeeler (license 325)
      
      (closes issue #18486)
      Reported by: bluefox
      Patches: 
            bug18486.patch uploaded by jpeeler (license 325)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d3c7a689
  51. Jan 24, 2011
    • Russell Bryant's avatar
      Merged revisions 303549 via svnmerge from · 09213439
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
  
  Merged revisions 303548 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
    
    Merged revisions 303546 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
      
      Fix channel redirect out of MeetMe() and other issues with channel softhangup.
      
      Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
      working properly.  This issue includes a patch that resolves the issue by
      removing a call to ast_check_hangup() from app_meetme.c.  I left that in my
      patch, as it doesn't need to be there.  However, the rest of the patch fixes
      this problem with or without the change to app_meetme.
      
      The key difference between what happens before and after this patch is the
      effect of the END_OF_Q control frame.  After END_OF_Q is hit in ast_read(),
      ast_read() will return NULL.  With the ast_check_hangup() removed, app_meetme
      sees this which causes it to exit as intended.  Checking ast_check_hangup()
      caused app_meetme to exit earlier in the process, and the target of the
      redirect saw the condition where ast_read() returned NULL.
      
      Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
      solve the issue if another application did the same thing.  There are also
      other edge cases where if an application finishes at the same time that a
      redirect happens, the target of the redirect will think that the channel hung
      up.  So, I made some changes in pbx.c to resolve it at a deeper level.  There
      are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
      abort the hangup process.  My patch extends this to remove the END_OF_Q frame
      from the channel's read queue, making the "abort hangup" more complete.  This
      same technique was used in every place where a softhangup flag was cleared.
      
      (closes issue #18585)
      Reported by: oej
      Tested by: oej, wedhorn, russell
      
      Review: https://reviewboard.asterisk.org/r/1082/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      09213439
  53. Jan 20, 2011