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  r166592 | tilghman | 2008-12-23 09:35:38 -0600 (Tue, 23 Dec 2008) | 3 lines
  
  Compile, even if both DAHDI and Zaptel are not installed.
  (Closes issue #14120)
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines

Fix a crash resulting from a datastore with inheritance but no duplicate callback

The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.

(closes issue #14113)
Reported by: francesco_r
Patches:
      14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r


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

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  r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines
  
  Use the integer form of condition for integer comparisons.
  (closes issue #14127)
   Reported by: andrew
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel.

(closes issue #13711)
Reported by: fmueller
Patches:
      13711.patch uploaded by putnopvut (license 60)



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

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r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines

Fix a deadlock relating to channel locks and autoservice

It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to 
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of 
autoservice.

The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.

The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.

In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.

It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.

(closes issue #14057)
Reported by: rtrauntvein
Patches:
      14057v3.patch uploaded by putnopvut (license 60)
Tested by: rtrauntvein

Review: http://reviewboard.digium.com/r/107/


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

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r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines

Fix up timeout handling in ast_carefulwrite().

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This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes.  It is currently used in AMI to
address the issue that has been reported.  However, there are probably a few
other places where this could be used.

(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13566)
Reported by: igorcarneiro
Tested by: russell
Review: http://reviewboard.digium.com/r/106/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166273 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port.
(closes issue #13628)
Reported by: pananix
Patches:
      bug13628.patch uploaded by jpeeler (license 325)
Tested by: file, blitzrage


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166268 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.

A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.

(closes issue #14118)
Reported by: blitzrage
Patches:
      14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage



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r166262 | russell | 2008-12-22 08:45:27 -0600 (Mon, 22 Dec 2008) | 7 lines

Re-work ref count handling of MoH classes using astobj2 to resolve crashes.

(closes issue #13566)
Reported by: igorcarneiro
Tested by: russell
Review: http://reviewboard.digium.com/r/106/

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This patch removes the usage of AST_PBX_KEEPALIVE from res_agi.  The only usage
was for the AGI command, "asyncagi break".  This patch removes this feature.
Normally, a feature would not be removed like this.  However, this code is
broken and usage of it will result in a memory leak.

Usage of this feature will make the AGI code return a result of 
AST_PBX_KEEPALIVE.  The PBX handler assumes that another thread has assumed
ownership of the channel.  The channel thread will exit without destroying the
channel.  Unfortunately, _no_ thread has ownership of the channel at this
point.  There are a couple of serious problems here:

1) The only way to recover the caller is to issue a channel redirect.  This
   will work, but this will be done with a masquerade, and the old ast_channel
   structure will be lost.

2) Until the channel redirect happens, there is no code servicing the channel.
   That means nothing is reading audio or handling events coming from the
   channel.  This is very bad.

The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code.  That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.


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I don't know how this crept back in when I had already
fixed it earlier



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r166157 | mmichelson | 2008-12-19 17:34:57 -0600 (Fri, 19 Dec 2008) | 9 lines

Backport of AUDIOHOOK_INHERIT for Asterisk 1.4

(closes issue #13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I could have sworn I already did that before, though...



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel 
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor 
continue recording the call even after the transfer
has completed.

It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.

(closes issue #13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
	  Tested by: putnopvut

Review: http://reviewboard.digium.com/r/102/



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r165991 | jpeeler | 2008-12-19 13:48:00 -0600 (Fri, 19 Dec 2008) | 15 lines

(closes issue #13480)
Reported by: tzafrir

Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug:

#ifdef HAVE_ZAPTEL
  fd = open("/dev/zap/pseudo", O_RDWR);
#else
  fd = open("/dev/dahdi/pseudo", O_RDWR);
#endif

is replaced with:
  fd = open(DAHDI_FILE_PSEUDO, O_RDRW);


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<value> tags inside <variable> elements must have CDATA and no
another XML node.


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

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r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines

Ensure that the chanspy datastore is fully initialized.

This patch resolved some random crash issues observed by a user on a BSD system

(closes issue #14111)
Reported by: ys
Patches:
      app_chanspy.c.diff uploaded by ys (license 281)

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This documents the recommended outline to use for commit message.  It also
covers information on special tags that can be used in commit messages, as well
as the template functionality that is available on bugs.digium.com.

Review: http://reviewboard.digium.com/r/96/


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

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r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines

Make ast_carefulwrite() be more careful.

This patch handles some additional cases that could result in partial writes
to the file description.  This was done to address complaints about partial
writes on AMI.

(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/

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Reported by: mika

Add ActionID response to ping if sent with request.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165798 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines
  
  Add mutexes around accesses to the IMAP library interface.  This prevents
  certain crashes, especially when shared mailboxes are used.
  (closes issue #13653)
   Reported by: howardwilkinson
   Patches: 
         asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590)
   Tested by: jpeeler
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(closes issue #13970)
Reported by: pkempgen
Patches:
      __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165792 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The variable "class" was being set NULL just prior to
being dereferenced in an ao2_link call. I have moved
the setting of the variable to NULL until after the
ao2_link call.



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This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source.  While this usage was perfectly safe,
there are others that are problematic.  Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.

Further changes to get rid of KEEPALIVE and related code is being done by
murf.  There is a patch up for that on review 29.

Review: http://reviewboard.digium.com/r/98/


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

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r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines

Set the process group ID on the MOH process so that all children will get killed

(closes issue #14099)
Reported by: caspy
Patches:
      res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645)

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

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  r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
  
  Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
  (closes issue #13545)
  Reported by: davidw
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  r165537 | file | 2008-12-18 12:07:37 -0400 (Thu, 18 Dec 2008) | 6 lines
  
  Do not crash if we are not passed in a followme id.
  (closes issue #14106)
  Reported by: ys
  Patches:
        app_followme.c.2.diff uploaded by ys (license 281)
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access 'struct ast_str' members, but this must only be used inside the API implementation.

(closes issue #14098)
Reported by: eliel
Patches:
      ast_str.patch uploaded by eliel (license 64)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165502 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Jared Smith suggested that we add a way to be able to set variables
and functions on the outbound channel.  I think that it's a great idea, so I
have added it as a todo so that it gets done at some point.


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