In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.



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  r191778 | mmichelson | 2009-05-02 13:48:20 -0500 (Sat, 02 May 2009) | 11 lines
  
  Fix a bug which resulted from the Hebrew voicemail commit.
  
  This fixes a case where a certain message could get played twice.
  
  (closes issue #13155)
  Reported by: greenfieldtech
  Patches:
        app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369)
  Tested by: greenfieldtech
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191781 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This feels like a sane change (wouldn't compile without this addition), but I'm
not intimately familiar with this code.


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  r191628 | mmichelson | 2009-05-02 05:21:00 -0500 (Sat, 02 May 2009) | 8 lines
  
  Move static buffers to outside for loops in app_chanspy.
  
  Similar to seanbright's commit 191422, this moves some static buffers
  to be defined outside of for loops since it is undefined if memory
  will be re-used or if the stack will grow with each iteration of the
  loop.
........
  r191629 | mmichelson | 2009-05-02 05:45:24 -0500 (Sat, 02 May 2009) | 3 lines
  
  Kevin has informed me that thi sort of thing is not necessary.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191630 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines
  
  SIP Response 410 maps to cause code 22 (or 23), not 1.
  (closes issue #14993)
   Reported by: BigJimmy
   Patches: 
         causepatch uploaded by BigJimmy (license 371)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15007)
 Reported by: hulber


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines
  
  Fix DTMF not being sent to other side after a partial feature match
  
  This fixes a regression from commit 176701. The issue was that
  ast_generic_bridge never exited after the feature digit timeout had elapsed,
  which prevented the queued DTMF from being sent to the other side.
  
  This issue was reported to me directly.
........


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........
  r191041 | seanbright | 2009-04-29 11:23:07 -0400 (Wed, 29 Apr 2009) | 6 lines
  
  Fix a crash in app_queue with very long member lists.
  
  A user reported via #asterisk that with very long lists of members, a crash
  occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead
  of stack allocating copys of each interface name.
........
  r191422 | seanbright | 2009-05-01 11:42:48 -0400 (Fri, 01 May 2009) | 7 lines
  
  Move the defintion of the a couple arrays out of loops.
  
  According to Kevin, it is unspecified as to whether a variable defined inside
  a block is allocated once by the compiler or for each pass through the block
  (loops being the only interesting case), so just define these before we get
  into our loop to be sure.
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191418 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reported by Andrew Lindh via the -dev list.


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If backgrounding and no core will be produced, then changing the directory
won't break anything; likewise, if the CWD isn't accessible by the current
user, then a core wasn't possible anyway.
(closes issue #14831)
 Reported by: chris-mac
 Patches: 
       20090428__bug14831.diff.txt uploaded by tilghman (license 14)
       20090430__bug14831.diff.txt uploaded by tilghman (license 14)
 Tested by: chris-mac


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

........
  r191220 | tilghman | 2009-04-29 18:10:54 -0500 (Wed, 29 Apr 2009) | 2 lines
  
  Allow H.323 to compile with FDLEAK checking enabled.
........


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chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified).  This patch lets the user pick the SSL/TLS client method for outbound connections in sip.

(closes issue #14770)
Reported by: TheOldSaint

(closes issue #14768)
Reported by: TheOldSaint

Review: http://reviewboard.digium.com/r/240/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For outgoing PTP redirected calls, you now need to use the inhibit(i)
option on all of the REDIRECTING statements before dialing the redirected-to
party.  You still have to set the REDIRECTING(to-xxx,i) and the
REDIRECTING(from-xxx,i) values.  The PTP call will update the redirecting-to
presentation when it becomes available and queue the redirecting update to
the calling channel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191175 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result.  No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/


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........
  r191096 | dbrooks | 2009-04-29 13:07:59 -0500 (Wed, 29 Apr 2009) | 8 lines
  
  Patch to fix tab-completion crash on "remove extension"
  
  This patch simply removes some old code back before Asterisk used editline. 
  This fixes the crash that occurred when tab-completing "remove extension".
  
  (closes issue #14689)
  Reported by: isaacgal
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files.  Before this change, SSL/TLS options were not consistent.  http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix.  While the options had different names in different conf files, they all did the exact same thing.  Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix.  For example.  'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files.  The change is noted in the CHANGES file though.

Review: http://reviewboard.digium.com/r/237/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14990)
Reported by: tzafrir
Patches:
      indications_err.diff uploaded by tzafrir (license 46)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I think it would behoove us to force "make validate-docs" to be run after the
XML documentation has been generated if dev-mode is enabled.

(closes issue #14989)
Reported by: tzafrir
Patches:
      app_queue_xml.diff uploaded by tzafrir (license 46)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190991 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue #14981)
Reported by: snuffy


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Change the build process so that doc/core-en_US.xml is dependent solely on the source files that have documentation in them, not on all source files.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190865 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them.



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*  Wait for a DivertingLegInformation3 message after receiving a
DivertingLegInformation1 message to complete the redirecting-to information
before queuing a redirecting update to the other channel.

*  A DivertingLegInformation2 message should be responded to with a
DivertingLegInformation3 when the COLR is determined.  If the call
could or does experience another redirection, you should manually
determine the COLR to send to the switch by setting REDIRECTING(to-pres)
to the COLR and setting REDIRECTING(to-num) = ${EXTEN}.

*  A DivertingLegInformation2 message must have an original called number
if the redirection count is greater than one.  Since Asterisk does
not keep track of this information, we can only indicate that the
number is not available due to interworking.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14979)
 Reported by: pj


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

........
  r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
  
  Fix 'inconsistent line endings' when autoconf 2.63 is used
  
  Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
  
  This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
........


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