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20 authors
  1. May 03, 2011
  3. Apr 27, 2011
    • Matthew Nicholson's avatar
      Merged revisions 315894 via svnmerge from · e87639fc
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines
  
  Merged revisions 315893 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
    
    Merged revisions 315891 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
      
      Fix our compliance with RFC 3261 section 18.2.2.
      
      This change optimizes the free_via() function and removes some redundant null
      checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
      the port specified in the Via header for routing responses (even when maddr is
      not set). Also the htons() function is now used when setting the port.
      Additional documentation comments have been added in various places to make the
      logic in the code clearer.
      
      (closes issue #18951)
      Reported by: jmls
      Patches:
            issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e87639fc
  5. Apr 26, 2011
  7. Apr 25, 2011
  9. Apr 22, 2011
  11. Apr 21, 2011
  13. Apr 18, 2011
    • Richard Mudgett's avatar
      Problems with ISDN MWI to phones. · 37274c73
      Richard Mudgett authored 
      The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself.  This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box.  The controlling user number should be made configurable.

JIRA ABE-2738
JIRA SWP-2846


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      37274c73
    • David Vossel's avatar
      Merged revisions 314067 via svnmerge from · 642249c3
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines
  
  Remove the need for deadlock avoidance in chan_sip do_monitor.
  
  Deadlock avoidance between the sip pvt and the pvt->owner is
  very difficult.  Now that channel's are ao2 objects, this complication
  is no longer necessary.  It turns out the pvt's msg queue only
  exists because of deadlock avoidance (when deadlock avoidance fails
  msgs were added to a queue to be processed later), so this goes away as well.
  
  The technique used in the new sip_lock_pvt_full() function should
  be used as a template for replacing all locations where deadlock
  avoidance occurs between a channel tech_pvt and the pvt's owner.
  My hope is that this will begin a reversal of the invalid channel
  driver locking architecture we have been using for so long. 
  
  This patch also resolves an issue where the pvt->owner gets
  unlocked during processing the msg queue.
  
  (closes issue #18690)
  Reported by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/1182/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      642249c3
    • David Vossel's avatar
      Merged revisions 314017 via svnmerge from · 4b454910
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
  
  sip codec negotiation of dynamic rtp payloads error fix
  
  This patch fixes how chan_sip handles dynamic rtp payload types
  it does not understand.  At the moment if a dynamic payload's mime
  type does not match one we understand, the payload does not get
  removed from our payload table.  As a result of this, the payload
  is set to whatever dynamic codec we use internally for that payload
  number on outgoing INVITES.  This is incorrect.
  
  This patch fixes this by properly checking the rtpmap set function's
  return code to make sure it was found.  The function can return both
  -1 and -2 depending on the source of the mismatch.  We were just
  checking -1 explicitly.
  
  Review: https://reviewboard.asterisk.org/r/1169/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4b454910
  15. Apr 17, 2011
  17. Apr 14, 2011
    • Richard Mudgett's avatar
      Merged revisions 313780 via svnmerge from · 4f8d56a8
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines
  
  Leftover debug messages unconditionally sent to the console.
  
  Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
  option enabled outputs the following debug messages unconditionally:
  
  Dialing T1847555121 on 1
  Dialing www2w on 1
  
  * Made debug messages in my_dial_digits() normal debug messages that do
  not get output unless enabled.
  
  * Reworded some debug messages in my_dial_digits() to be clearer.
  
  * Replace strncpy() with ast_copy_string() in my_dial_digits() which does
  the same job better.
  
  (closes issue #18847)
  Reported by: vmikhelson
  Tested by: rmudgett
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4f8d56a8
  19. Apr 13, 2011
  21. Apr 12, 2011
  23. Apr 11, 2011
    • Richard Mudgett's avatar
      Merged revisions 313190 via svnmerge from · bc907695
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines
  
  Merged revisions 313189 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines
    
    Merged revisions 313188 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines
      
      Stuck channel using FEATD_MF if caller hangs up at the right time.
      
      The cause was actually a caller hanging up just at the end of the Feature
      Group D DTMF tones that setup the call.  The reason for this is a "guard
      timer" that's implemented using ast_safe_sleep(100).  If the caller
      happens to hang up AFTER the final tone of the DTMF string but BEFORE the
      end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
      This causes the code to bounce to the end of ss_thread(), but it does NOT
      tear down the call properly.
      
      This should be a rare occurrence because the caller has to hang up at
      EXACTLY the right time.  Nonetheless, it was happening quite regularly on
      the reporter's system.  It's not easily reproducible, unless you purposely
      increase the guard-time to 2000 or more.  Once you do that, you can
      reproduce it every time by watching the DTMF debug and hanging up just as
      it ends.
      
      Simply add an ast_hangup() before goto quit.
      
      (closes issue #15671)
      Reported by: jcromes
      Patches:
            issue15671.patch uploaded by pabelanger (license 224)
      Tested by: jcromes
    ........
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      bc907695
  25. Apr 08, 2011
  27. Apr 07, 2011
  29. Apr 05, 2011
  31. Apr 04, 2011
    • Richard Mudgett's avatar
      Remove the channel parameter from sig_pri_handle_subcmds(). · 121b90a4
      Richard Mudgett authored 
      It was only used in a debug message and may not be correct anyway.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      121b90a4
    • Richard Mudgett's avatar
      Merged revisions 312575 via svnmerge from · e1ceb52b
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines
  
  Merged revisions 312574 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
    
    Merged revisions 312573 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
      
      Issues with ISDN calls changing B channels during call negotiations.
      
      The handling of the PROCEEDING message was not using the correct call
      structure if the B channel was changed.  (The same for PROGRESS.) The call
      was also not hungup if the new B channel is not provisioned or is busy.
      
      * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
      PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
      using the correct structure and B channel.  If there is any problem with
      the operations then the call is now hungup with an appropriate cause code.
      
      * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
      correct structure by looking for the call and not using the channel ID.
      NOTIFY is an exception with versions of libpri before v1.4.11 because a
      call pointer is not available for Asterisk to use.
      
      * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
      the correct structure by looking for the call and not using the channel
      ID.
      
      (closes issue #18313)
      Reported by: destiny6628
      Tested by: rmudgett
      JIRA SWP-2620
      
      (closes issue #18231)
      Reported by: destiny6628
      Tested by: rmudgett
      JIRA SWP-2924
      
      (closes issue #18488)
      Reported by: jpokorny
      JIRA SWP-2929
      
      JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
      JIRA DAHDI-406
      JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
    ........
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e1ceb52b
  33. Apr 01, 2011
  35. Mar 31, 2011
    • Richard Mudgett's avatar
      Merged revisions 312022 via svnmerge from · ee44bf72
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines
  
  chan_misdn segfaults when DEBUG_THREADS is enabled.
  
  The segfault happens because jb->mutexjb is uninitialized from the
  ast_malloc().  The internals of ast_mutex_init() were assuming a nonzero
  value meant mutex tracking initialization had already happened.  Recent
  changes to mutex tracking code to reduce excessive memory consumption
  exposed this uninitialized value.
  
  Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc().
  Also eliminated redundant zero initialization code in the routine.
  
  (closes issue #18975)
  Reported by: irroot
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ee44bf72
  37. Mar 30, 2011