Commit Graph

3268 Commits

Author SHA1 Message Date
Steve Murphy
879326b3f3 Merged revisions 174370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) | 10 lines
  
  Merged revisions 174369 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines
    
    This patch solves some compiler complaints
    in both 32 and 64-bit environments.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@174428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 04:09:42 +00:00
David Vossel
200935605b Merged revisions 174325 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174325 | dvossel | 2009-02-09 11:26:02 -0600 (Mon, 09 Feb 2009) | 9 lines
  
  Fixes issue with hangups not being sent and external process never terminating. 
  
  The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued.  If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. 
  
  (closes issue #14251)
  Reported by: chris-mac
  Tested by: dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@174330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 17:47:27 +00:00
Joshua Colp
993d0af118 Merged revisions 173902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb 2009) | 4 lines
  
  Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
  (closes issue #14414)
  Reported by: bluecrow76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 16:01:01 +00:00
Mark Michelson
41859bdb73 Merged revisions 173773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines

Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage

(closes issue #13905)
Reported by: jaroth
Patches:
      foldermove_v2.patch uploaded by jaroth (license 50)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 23:31:53 +00:00
Jeff Peeler
9f3e4e86b3 Merged revisions 173697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines
  
  Merged revisions 173696 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
    
    Add new configuration option to make shared IMAP mailboxes function as expected.
    
    The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
    using the same IMAP storage location to function as one mailbox. This allows
    all messages to be retrieved for any user in the group. The patch alters the
    'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
    for a given user.
    
    (closes issue #13673)
    Reported by: howardwilkinson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 21:06:12 +00:00
Mark Michelson
d9694895c9 Merged revisions 173693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines

Merged revisions 173692 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines

Fix situations where queue members could be autopaused unexpectedly

Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.

(closes issue #14376)
Reported by: fiddur
Patches:
      14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 20:35:42 +00:00
Mark Michelson
075fb9b0cd Merged revisions 173593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines

Merged revisions 173592 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines

Add some missing cleanup to app_mixmonitor


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 18:50:00 +00:00
Mark Michelson
616a6fdd54 Merged revisions 173589 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines

Merged revisions 173559 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines

Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.

app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).

The solution for this is to employ a datastore, which has the nice benefit of allowing us 
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.

app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!

(closes issue #14374)
Reported by: aragon
Patches:
	  14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 18:42:12 +00:00
Mark Michelson
e24a4644e4 Merged revisions 173507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines

Fix some areas where the incorrect interface was passed to ast_device_state

I swear it feels like I already did this once...

(closes issue #14359)
Reported by: francesco_r

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 22:25:06 +00:00
Mark Michelson
f932b09c5f Merged revisions 173397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines

Merged revisions 173396 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines

Revert my previous change because it was stupid


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 17:46:36 +00:00
Mark Michelson
0017b5cf52 Merged revisions 173393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines

Merged revisions 173392 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines

Add a missing unlock. Extremely unlikely to ever matter, but it's needed.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@173395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 17:42:41 +00:00
Tilghman Lesher
13508cc9f3 Merged revisions 172741 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines
  
  Blank argument crashes Asterisk
  (closes issue #14377)
   Reported by: amorsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@172743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-01 02:45:47 +00:00
Terry Wilson
ef566503d3 Merged revisions 172580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines
  
  Merged revisions 172517 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
    
    Fix feature inheritance with builtin features
    
    When using builtin features like parking and transfers, the AST_FEATURE_* flags
    would not be set correctly for all instances when either performing a builtin
    attended transfer, or parking a call and getting the timeout callback.  Also,
    there was no way on a per-call basis to specify what features someone should
    have on picking up a parked call (since that doesn't involve the Dial() command).
    There was a global option for setting whether or not all users who pickup a
    parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
    AUTOMON, or PARKCALL.
    
    This patch:
    1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
    dialplan or with setvar in channels that support it.  This variable can be set
    to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
    equivalent dial options), to set what features should be activated on this
    channel.  The patch moves the setting of the features datastores into the
    bridging code instead of app_dial to help facilitate this.
    
    2) adds global options parkedcallparking, parkedcallhangup, and
    parkedcallrecording to be similar to the parkedcalltransfers option for
    globally setting features.
    
    3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
    extension since tracking everything through multiple masquerades, etc. is
    difficult and error-prone
    
    4) attempts to fix all cases of return calls from parking and completed builtin
    transfers not having the correct permissions
    (closes issue #14274)
    Reported by: aragon
    Patches: 
          fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
    Tested by: aragon, otherwiseguy
    
    Review http://reviewboard.digium.com/r/138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@172636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-31 00:05:17 +00:00
Tilghman Lesher
95c892e3f9 Merged revisions 172441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines
  
  Merged revisions 172438 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
    
    Lose the CAP_NET_ADMIN at every fork, instead of at startup.  Otherwise, if
    Asterisk runs as a non-root user and the administrator does a 'restart now',
    Asterisk loses the ability to set QOS on packets.
    (closes issue #14004)
     Reported by: nemo
     Patches: 
           20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Corydon76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@172504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 23:53:12 +00:00
Steve Murphy
dfb881fadf Merged revisions 172063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines
  
  Merged revisions 172030 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
    
    This patch fixes h-exten running misbehavior in manager-redirected 
    situations.
    
    What it does:
    1. A new Flag value is defined in include/asterisk/channel.h,
     AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
     bridge hangup exten code not to run the h-exten there (nor
     publish the bridge cdr there). It will done at the pbx-loop
     level instead.
    2. In the manager Redirect code, I set this flag on the channel
     if the channel has a non-null pbx pointer. I did the same for the
     second (chan2) channel, which gets run if name2 is set...
     and the first succeeds.
    3. I restored the ending of the cdr for the pbx loop h-exten
     running code. Don't know why it was removed in the first place.
    4. The first attempt at the fix for this bug was to place code
       directly in the async_goto routine, which was called from a
       large number of places, and could affect a large number of
       cases, so I tested that fix against a fair number of transfer
       scenarios, both with and without the patch. In the process,
       I saw that putting the fix in async_goto seemed not to affect
       any of the blind or attended scenarios, but still, I was
       was highly concerned that some other scenarios I had not tested
       might be negatively impacted, so I refined the patch to 
       its current scope, and jmls tested both. In the process, tho,
       I saw that blind xfers in one situation, when the one-touch
       blind-xfer feature is used by the peer, we got strange 
       h-exten behavior.  So, I  inserted code to swap CDRs and
       to set the HANGUP_DONT field, to get uniform behavior.
    5. I added code to the bridge to obey the HANGUP_DONT flag,
       skipping both publishing the bridge CDR, and running
       the h-exten; they will be done at the pbx-loop (higher)
       level instead.
    6. I removed all the debug logs from the patch before committing.
    7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
       so it's only done if the h-exten is going to be run. A very
       minor performance improvement, but technically correct.
    
    
    (closes issue #14241)
    Reported by: jmls
    Patches:
          14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
    Tested by: murf, jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@172067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28 20:56:48 +00:00
Mark Michelson
2d42d6cc64 Merged revisions 171618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines

Fix queue crashes that would occur after the calling channel was masqueraded.

The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.

To fix this problem, there are two measures taken

1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c

(closes issue #14260)
Reported by: ccesario
Patches:
      14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@171620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 19:43:22 +00:00
Sean Bright
3595adcf68 Merged revisions 170980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines
  
  Merged revisions 170979 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines
    
    Resolve a logic error that was causing Page() to crash when more than one
    channel was specified.
    
    (closes issue #14308)
    Reported by: bluefox
    Patches:
          20090124__bug14308.diff.txt uploaded by seanbright (license 71)
    Tested by: kc0bvu
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-25 13:40:52 +00:00
Joshua Colp
692b93004e Merged revisions 170569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170568 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.
    (closes issue #14310)
    Reported by: RadicAlish
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 19:10:43 +00:00
Joshua Colp
97cae952ea Merged revisions 170148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines
  
  Merged revisions 170147 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
    
    If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
    (closes issue #14282)
    Reported by: cheesegrits
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 16:54:45 +00:00
Joshua Colp
d5f88932f4 Merged revisions 170047 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines
  
  Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop.
  (closes issue #14304)
  Reported by: jcovert
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 15:04:44 +00:00
Mark Michelson
e2ce2d9fab Merged revisions 169611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines

Fix device state parsing issues for channel names with multiple slashes

The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.

In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.

(closes issue #14014)
Reported by: kebl0155
Patches:
      14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@169613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 00:35:44 +00:00
Mark Michelson
b072c19f40 Merged revisions 169574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r169574 | mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6 lines

Use the default timeout for a queue instead of -1

(closes issue #14272)
Reported by: timking


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@169575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 21:57:49 +00:00
Tilghman Lesher
895e42e525 Merged revisions 169365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r169365 | tilghman | 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) | 11 lines
  
  Merged revisions 169364 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines
    
    Truncate userevents at the end of a line, when the command exceeds the buffer.
    (closes issue #14278)
     Reported by: fnordian
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@169368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-19 20:10:13 +00:00
Tilghman Lesher
5b50a09ef2 Merged revisions 168832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines
  
  Merged revisions 168828 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines
    
    Fix the conjugation of Russian and Ukrainian languages.
    (related to issue #12475)
     Reported by: chappell
     Patches: 
           vm_multilang.patch uploaded by chappell (license 8)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@168836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 18:55:13 +00:00
Sean Bright
71ebf2e7ca Merged revisions 168705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines
  
  Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
  conferences.  We were using the 'user number' field to compare against the
  maximum allowed users, which works assuming users with lower user numbers
  didn't leave the conference.
  
  (closes issue #14117)
  Reported by: sergedevorop
  Patches:
        20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
  Tested by: sergedevorop
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2009-01-15 15:37:48 +00:00
Mark Michelson
fac78dd00e Merged revisions 168629 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines

Merged revisions 168628 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines

Fix some crashes from bad datastore handling in app_queue.c

* The queue_transfer_fixup function was searching for and removing
  the datastore from the incorrect channel, so this was fixed.

* Most datastore operations regarding the queue_transfer datastore
  were being done without the channel locked, so proper channel locking
  was added, too.

(closes issue #14086)
Reported by: ZX81
Patches:
      14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr


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2009-01-15 00:15:22 +00:00
Steve Murphy
e122dcc43f Merged revisions 168613 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9 lines
  
  Merged revisions 168608 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line
    
    app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning.
  ........
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2009-01-14 21:30:36 +00:00
Terry Wilson
48639b1604 Merged revisions 168594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines
  
  Merged revisions 168593 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
    
    Don't overflow when paging more than 128 extensions
    
    The number of available slots for calls in app_page was hardcoded to 128.
    Proper bounds checking was not in place to enforce this limit, so if more than
    128 extensions were passed to the Page() app, Asterisk would crash.  This patch
    instead dynamically allocates memory for the ast_dial structures and removes
    the (non-functional) arbitrary limit.
    
    This issue would have special importance to anyone who is dynamically creating
    the argument passed to the Page application and allowing more than 128
    extensions to be added by an outside user via some external interface.
    
    The patch posted by a_villacis was slightly modified for some coding guidelines
    and other cleanups.  Thanks, a_villacis!
    (closes issue #14217)
    Reported by: a_villacis
    Patches: 
          20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
    Tested by: otherwiseguy
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2009-01-14 02:11:00 +00:00
Russell Bryant
a29672060a Merged revisions 168562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines

Merged revisions 168561 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines

Revert unnecessary indications API change from rev 122314

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2009-01-13 19:35:59 +00:00
Olle Johansson
b1a8c36bb0 Merged revisions 168497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168497 | oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines

Better to use the proper app name

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2009-01-12 17:26:38 +00:00
Terry Wilson
a671c82ec2 Merged revisions 167935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines
  
  Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set
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2009-01-09 00:45:37 +00:00
Tilghman Lesher
361d76c1eb Merged revisions 167835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167835 | tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines
  
  Textual changes, consistency in status variable naming, and other minor bugs.
  (closes issue #13943)
   Reported by: Marquis
   Patches: 
         minivm_trunk_fixes3.patch uploaded by Marquis (license 32)
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2009-01-08 21:39:31 +00:00
BJ Weschke
32be29070a Merged revisions 167478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167478 | bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines
  
   Answer the channel if it has not already been answered and we've already found a valid profile for followme. 
   (closes issue #14140)
   Reported by: dimas
   Patches:
         14140.patch uploaded by dimas
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2009-01-07 18:32:52 +00:00
Mark Michelson
45ac5e6838 Merged revisions 166861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines

Update app_queue to deal with the removal of AST_PBX_KEEPALIVE

When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.

I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.


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2008-12-29 18:16:10 +00:00
Steve Murphy
a7aeaf341b Merged revisions 166665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

This merged from trunk with no conflicts. I tested
mostly the 'tired' cases, and for the most part
ignored the tests for reconnecting and dialing in
to fetch a parked call, after the first case.

................
  r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines
  
  Merged revisions 166093 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  In order to merge this 1.4 patch into trunk,
  I had to resolve some conflicts and wait for
  Russell to make some changes to res_agi.
  I re-ran all the tests; 39 calls in all, and
  made fairly careful notes and comparisons: I
  don't want this to blow up some aspect of 
  asterisk; I completely removed the KEEPALIVE
  from the pbx.h decls. The first 3 scenarios
  involving feature park; feature xfer to 700;
  hookflash park to Park() app call all behave
  the same, don't appear to leave hung channels,
  and no crashes.
  
  ........
    r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
    
    This merges the masqpark branch into 1.4
    
    These changes eliminate the need for (and use of)
    the KEEPALIVE return code in res_features.c;
    There are other places that use this result code
    for similar purposes at a higher level, these appear
    to be left alone in 1.4, but attacked in trunk.
    
    The reason these changes are being made in 1.4, is
    that parking ends a channel's life, in some situations,
    and the code in the bridge (and some other places),
    was not checking the result code properly, and dereferencing
    the channel pointer, which could lead to memory corruption
    and crashes.
    
    Calling the masq_park function eliminates this danger 
    in higher levels.
    
    A series of previous commits have replaced some parking calls
    with masq_park, but this patch puts them ALL to rest,
    (except one, purposely left alone because a masquerade
    is done anyway), and gets rid of the code that tests
    the KEEPALIVE result, and the NOHANGUP_PEER result codes.
    
    While bug 13820 inspired this work, this patch does
    not solve all the problems mentioned there.
    
    I have tested this patch (again) to make sure I have
    not introduced regressions. 
    
    Crashes that occurred when a parked party hung up
    while the parking party was listening to the numbers
    of the parking stall being assigned, is eliminated.
    
    These are the cases where parking code may be activated:
    
    1. Feature one touch (eg. *3)
    2. Feature blind xfer to parking lot (eg ##700)
    3. Run Park() app from dialplan (eg sip xfer to 700)
       (eg. dahdi hookflash xfer to 700)
    4. Run Park via manager.
    
    The interesting testing cases for parking are:
    I. A calls B, A parks B
        a. B hangs up while A is getting the numbers announced.
        b. B hangs up after A gets the announcement, but 
           before the parking time expires
        c. B waits, time expires, A is redialed,
           A answers, B and A are connected, after
           which, B hangs up.
        d. C picks up B while still in parking lot.
    
    II. A calls B, B parks A
        a. A hangs up while B is getting the numbers announced.
        b. A hangs up after B gets the announcement, but 
           before the parking time expires
        c. A waits, time expires, B is redialed,
           B answers, A and B are connected, after
           which, A hangs up.
        d. C picks up A while still in parking lot.
    
    Testing this throroughly involves acting all the permutations
    of I and II, in situations 1,2,3, and 4.
    
    Since I added a few more changes (ALL references to KEEPALIVE in the bridge
    code eliimated (I missed one earlier), I retested
    most of the above cases, and no crashes.
    
    H-extension weirdness.
    
    Current h-extension execution is not completely
    correct for several of the cases.
    
    For the case where A calls B, and A parks B, the
    'h' exten is run on A's channel as soon as the park
    is accomplished. This is expected behavior.
    
    But when A calls B, and B parks A, this will be
    current behavior:
    
    After B parks A, B is hung up by the system, and
    the 'h' (hangup) exten gets run, but the channel
    mentioned will be a derivative of A's...
    
    Thus, if A is DAHDI/1, and B is DAHDI/2,
    the h-extension will be run on channel
    Parked/DAHDI/1-1<ZOMBIE>, and the 
    start/answer/end info will be those 
    relating to Channel A.
    
    And, in the case where A is reconnected to
    B after the park time expires, when both parties
    hang up after the joyful reunion, no h-exten
    will be run at all.
    
    In the case where C picks up A from the 
    parking lot, when either A or C hang up,
    the h-exten will be run for the C channel.
    
    CDR's are a separate issue, and not addressed
    here.
    
    As to WHY this strange behavior occurs, 
    the answer lies in the procedure followed
    to accomplish handing over the channel
    to the parking manager thread. This procedure
    is called masquerading. In the process,
    a duplicate copy of the channel is created,
    and most of the active data is given to the
    new copy. The original channel gets its name
    changed to XXX<ZOMBIE> and keeps the PBX
    information for the sake of the original
    thread (preserving its role as a call 
    originator, if it had this role to begin
    with), while the new channel is without
    this info and becomes a call target (a
    "peer").
    
    In this case, the parking lot manager
    thread is handed the new (masqueraded)
    channel. It will not run an h-exten
    on the channel if it hangs up while
    in the parking lot. The h exten will
    be run on the original channel instead,
    in the original thread, after the bridge
    completes.
    
    See bug 13820 for our intentions as
    to how to clean up the h exten behavior.
  
  Review: http://reviewboard.digium.com/r/29/
  
  ........
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2008-12-24 01:15:43 +00:00
Russell Bryant
e3f0577bda Merged revisions 165890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines

Merged revisions 165889 via svnmerge from 
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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2008-12-19 15:08:59 +00:00
Tilghman Lesher
a3825a040a Merged revisions 165797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165797 | tilghman | 2008-12-18 15:41:02 -0600 (Thu, 18 Dec 2008) | 15 lines
  
  Merged revisions 165767 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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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2008-12-18 21:57:14 +00:00
Joshua Colp
4142e300f7 Merged revisions 165792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines
  
  Numerous documentation updates.
  (closes issue #13970)
  Reported by: pkempgen
  Patches:
        __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10)
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2008-12-18 21:24:00 +00:00
Russell Bryant
52f5e8fcfb Merged revisions 165723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines

Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.

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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2008-12-18 19:45:11 +00:00
Tilghman Lesher
fcb9c545f4 Merged revisions 165658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines
  
  Fix 2 resource leaks and fix another pipe-to-comma conversion
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2008-12-18 18:47:08 +00:00
Tilghman Lesher
d70704c277 Merged revisions 165325 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165325 | tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines
  
  Oops, broke trunk
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2008-12-17 21:31:15 +00:00
Mark Michelson
04b50b8b8c Merged revisions 165318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines

Merged revisions 165255 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines

Fix some memory leaks found while looking at how realtime
configs are handled.

Also cleaned up some coding guidelines violations in app_realtime.c,
mostly related to spacing


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2008-12-17 21:25:32 +00:00
Tilghman Lesher
c383e518fd Merged revisions 165319 via svnmerge from
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  r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) | 11 lines
  
  Merged revisions 165317 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines
    
    Reverse the fix from issue #6176 and add proper handling for that issue.
    (Closes issue #13962, closes issue #13363)
    Fixed by myself (license 14)
  ........
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2008-12-17 21:22:38 +00:00
Mark Michelson
5f50fb1dce Merged revisions 165142-165143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, 17 Dec 2008) | 10 lines

Use the create_vm_state_from_user function in a place where
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.

This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.


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r165143 | mmichelson | 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines

And actually assign the function to a pointer...


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2008-12-17 17:56:53 +00:00
Jeff Peeler
8cbe3fd12d Merged revisions 164942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164942 | jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines

(closes issue #13669)
Reported by: pj

Delete file recording if recording terminated from a hangup.


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2008-12-16 22:52:17 +00:00
Russell Bryant
57bb6ea061 Merged revisions 164877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines

Merged revisions 164876 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines

Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.

This is a bug I noticed while looking at the code for app_macro.  This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched.  (I hate this return code with a passion, by the way.)

........

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2008-12-16 21:13:53 +00:00
Russell Bryant
8d40c280fd Merged revisions 164623 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164623 | russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines

Set MINIVM_ACCMESS_STATUS in all cases.  Also, remove a variable that was not needed.

(closes issue #14081)
Reported by: pkempgen

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16 15:02:14 +00:00
Mark Michelson
e539850d4e Merged revisions 164270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164270 | mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4 lines

Fix a compile warning and a logic error that could have been bad
for non-realtime queues


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 16:18:38 +00:00
Mark Michelson
20f7c5cb05 Merged revisions 164268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines

Fix up a few issues with regards to queues

* Fix reference counting used in the __queues_show function
* Add code to be sure that the "queue show" command does not
  print information for a realtime queue which has been deleted
  from the backend
* Add a missing unref to the realtime queue loading function for
  the case where a queue is in the module's container but has been
  deleted from the realtime backend

(closes issue #14033)
Reported by: cristiandimache
Patches:
      14033.patch uploaded by putnopvut (license 60)
Tested by: cristiandimache


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 16:18:19 +00:00
Joshua Colp
054e20548b Merged revisions 164257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r164257 | file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines
  
  Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret.
  (closes issue #14073)
  Reported by: seandarcy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 15:50:12 +00:00