https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines
  
  Make sure to recreate the dahdi pseudo channel after dahdi restart
  
  (closes issue #14477)
  Reported by: timking
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The original patch for this was written by Brett Bryant, and I split it out into
it's own module.

(closes issue #12876)
Reported by: bbryant
Patches:
      06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
      05212009_cdr_syslog.patch uploaded by seanbright (license 71)
Tested by: seanbright

Review: https://reviewboard.asterisk.org/r/297/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203846 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14594)
Reported by: JimDickenson
Patches:
      chanspy.diff uploaded by JimDickenson (license 710)


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines
  
  Don't fast forward past the end of a message.
  
  This is nice change for users of the voicemail application.  If someone gets a
  little carried away with fast forwarding through a message, they can easily
  get to the end and accidentally exit the voicemail application by hitting the
  fast forward key during the following prompt.
  
  This adds some safety by not allowing a fast forward past the end of a message.
  
  (closes issue #14554)
  Reported by: lacoursj
  Patches:
        21761.patch uploaded by lacoursj (license 707)
  Tested by: lacoursj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203802 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14596)
Reported by: JimDickenson
Patches:
      pong2.diff uploaded by JimDickenson (license 710)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14452)
Reported by: umberto71


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203779 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix the 'nat' option to actually do RFC3581 as expected and extend the configurable values for finer control.

(closes issue #8855)
Reported by: mikma
Tested by: klaus3000, file


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
as seconds.

Also, the inequality was reversed. The warning, if triggered, was "Max silence should 
be less than minmessage or you may get empty messages", which should have been logged 
if max silence was greater than minmessage, but the check was for less than.

Also, conforming if statement to coding guidelines.

closes issue #15331)
Reported by: markd

Review: https://reviewboard.asterisk.org/r/293/


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  r203719 | dbrooks | 2009-06-26 15:03:42 -0500 (Fri, 26 Jun 2009) | 16 lines
  
  Fixing voicemail's error in checking max silence vs min message length
  
  Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
  as seconds.
  
  Also, the inequality was reversed. The warning, if triggered, was "Max silence should 
  be less than minmessage or you may get empty messages", which should have been logged 
  if max silence was greater than minmessage, but the check was for less than.
  
  Also, conforming if statement to coding guidelines.
  
  closes issue #15331)
  Reported by: markd
  
  Review: https://reviewboard.asterisk.org/r/293/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203720 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15404)
Reported by: leobrown
Patches:
      iax_codec_debug.patch uploaded by leobrown (license 541)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14413)
Reported by: pj


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after a polarity reversal.

Previously on a polarity switch event chan_dahdi would set the channel
immediately as answered. This would cause problems if a polarity reversal
occurred when the line was picked up as the dial would not have yet occurred. 
Now if the polarity reversal occurs before delay has elapsed after coming off
hook or an answer, it is ignored. Also, some refactoring was done in
_handle_event.

(closes issue #13917)
Reported by: alecdavis
Patches:
      chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis


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CEL is the new system for logging channel events.  This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records.  For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.

Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code.  Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.

Review: https://reviewboard.asterisk.org/r/239/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also change the default casing of the string contants to lowercase.  This really
just saves us from have to lowercase them later when displaying them.


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This has the pleasant side effect of cleaning up the header inclusion process
in logger.c.


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(issue #15269)


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines
  
  I didn't see that Mark already fixed the underlying issue!
  
  Yay for removing useless code.
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines
  
  Fix a case where CDR answer time could be before the start time involving parking.
  
  (closes issue #13794)
  Reported by: davidw
  Patches:
        13794.patch uploaded by murf (license 17)
        13794.patch.160 uploaded by murf (license 17)
  Tested by: murf, dbrooks
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines
  
  Don't try to free NULL
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
sig_pri.c. Functionality in theory should not change (mostly). A few trivial
changes were made in sig_analog with verbose messages and commenting.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This would occasionally cause one-way audio when using hardware DTMF detection.

(closes issue #14761)
Reported by: tzafrir
Patches:
      v1-14761.patch uploaded by dimas (license 88)
Tested by: tzafrir, dimas


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........
  r203230 | mmichelson | 2009-06-25 13:52:22 -0500 (Thu, 25 Jun 2009) | 3 lines
  
  Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled.
........


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(closes issue #11797)
Reported by: macbrody

Review: https://reviewboard.asterisk.org/r/270/


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Moved SETCADENCE ioctl call to before call into new analog signal module
to insure that it gets set. 

(closes issue #15381)
Reported by: alecdavis
Patches:
      fix15381.diff uploaded by dbailey (license 819)
Tested by: dbailey



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
  
  Resolve a crash related to a T.38 reinvite race condition.
  
  This change resolves a crash observed locally during some T.38 testing.
  A call was set up using a call file, and when the T.38 reinvite came in,
  the channel state was still AST_STATE_DOWN.  The reason is explained by
  a comment in the code that previously lived in the handling of
  AST_STATE_RINGING.  This change modifies the logic to handle the same
  race condition for any channel state that is not UP.
  
  (closes ABE-1895)
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
  
  Improved chan_dahdi.conf pritimer error checking.
  
  Valid format is: pritimer=timer_name,timer_value
  
  *  Fixed segfault if the ',' is missing.
  *  Completely check the range returned by pri_timer2idx() to prevent
  possible access outside array bounds.
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines
  
  Use the handy UNLINK macro instead of hand-coding the same thing in-line.
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Also change the preferred configuration option from 'hostname' (which was
misleading because it didn't actually treat the value as a hostname) to
'connection' and added some verbage explaining that the user would need to
refer to their freetds.conf file for those settings.  'hostname' was kept
as a backwards compatible configuration parameter.


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