Commit graph

480 commits

Author SHA1 Message Date
Jonathan Rose
c0b9fe8530 Send relative path named recordings to the meetme directory instead of sounds
Prior to this patch, no effort was made to parse the path name to determine a proper
destination for recordings of MeetMe's r option. This fixes that.

Review: https://reviewboard.asterisk.org/r/1846/
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Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362080 from http://svn.asterisk.org/svn/asterisk/branches/10


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2012-04-13 15:38:08 +00:00
Matthew Jordan
aa21d4fc6b Fix memory leak when using MeetMeAdmin 'e' option with user specified
A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command
(eject last user that joined) is used in conjunction with a specified user.
Regardless of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user.  Because the 'e' option kicks
the last user that joined, as opposed to the one specified, the reference to
the user specified by the command would be leaked when the user variable
was assigned to the last user that joined.
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Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2012-04-06 20:32:52 +00:00
Jonathan Rose
fc45af331b Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined
There were a few instances of restarting music on hold in meetme that would cause
Asterisk to revert to the default class of music on hold for no adequate reason.

Review: https://reviewboard.asterisk.org/r/1844/
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Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2012-04-05 17:22:30 +00:00
Jonathan Rose
97b2fa8de1 Make the MeetMeAdmin N command (mute all nonadmins) not mute admins
(Closes Issue ASTERISK-19335)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1843/
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Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2012-04-04 13:51:45 +00:00
Terry Wilson
0e5c761c28 Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/


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2012-03-01 22:09:18 +00:00
Terry Wilson
a9d607a357 Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/


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2012-02-29 16:52:47 +00:00
Terry Wilson
ebaf59a656 Opaquification for ast_format structs in struct ast_channel
Review: https://reviewboard.asterisk.org/r/1770/


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2012-02-24 00:32:20 +00:00
Terry Wilson
57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


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2012-02-20 23:43:27 +00:00
Terry Wilson
34c55e8e7c Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/


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2012-02-13 17:27:06 +00:00
Joshua Colp
afdd96712c Make the 'c' option to MeetMe work even if the 'q' option is used.
(closes issue ASTERISK-17053)
Reported by: justdave


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2012-02-06 16:38:23 +00:00
Terry Wilson
99cae5b750 Opaquify channel stringfields
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.

Review: https://reviewboard.asterisk.org/r/1661/


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2012-01-24 20:12:09 +00:00
Terry Wilson
04da92c379 Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


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2012-01-09 22:15:50 +00:00
Kinsey Moore
c04f4d72fd Prevent SLA settings from getting wiped out on reload
If SLA was reloaded without the config file being changed, current settings got
wiped out before the SLA reload code decided it wasn't going to reload the file
since nothing was changed.  Moving the settings reset later in the reload
process fixes this.

(closes issue AST-744)
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Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2012-01-09 15:40:16 +00:00
Jonathan Rose
8e94432d9a Fix: Meetme recording variables from realtime DB use null entries over channel variables
Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.

(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
	ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
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Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2011-12-07 20:34:23 +00:00
Kinsey Moore
dc05ce5e4f Fix another incorrect case with meetme's PIN logic and add documentation
This fixes an issue where a user of a dynamic conference was asked for a PIN
twice.  This also adds documentation to assist in future modifications to the
piece of code responsible for PIN checking.

(closes issue AST-670)
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Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2011-11-10 21:15:39 +00:00
Kinsey Moore
c1647ab33a Fix pin parameter behavior regression in MeetMe
The last time this code was touched (by me), a subtlety was missed based on the
difference between needing to check a pin's validity and the need to prompt
for a pin.

(closes issue ASTERISK-18488)
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Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2011-11-09 17:15:44 +00:00
Kevin P. Fleming
784bbf70d7 Modify comments in MeetMe application documentation about DAHDI.
The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
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2011-11-02 13:46:15 +00:00
Richard Mudgett
796ed62f47 Update MeetMe p and X option documentation when interacting with the s option.
ASTERISK-12175 changed the p and X options to not interfere with the s
option when they are used together.  It makes more sense for the s option
to have priority for the DTMF '*' key since it cannot change its
activation code.  Otherwise, you could not use option s with the p or X
options.

JIRA AST-671
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2011-10-12 17:52:55 +00:00
Matthew Jordan
e218748ac1 Merged revisions 337120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines
  
  Merged revisions 337118 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines
    
    Fix for incorrect voicemail duration in external notifications
    
    This patch fixes an issue where the voicemail duration was being reported
    with a duration significantly less than the actual sound file duration.
    Voicemails that contained mostly silence were reporting the duration of
    only the sound in the file, as opposed to the duration of the file with
    the silence.  This patch fixes this by having two durations reported in
    the __ast_play_and_record family of functions - the sound_duration and the
    actual duration of the file.  The sound_duration, which is optional, now
    reports the duration of the sound in the file, while the actual full duration
    of the file is reported in the duration parameter.  This allows the voicemail
    applications to use the sound_duration for minimum duration checking, while
    reporting the full duration to external parties if the voicemail is kept.
    
    (issue ASTERISK-2234)
    (closes issue ASTERISK-16981)
    Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
    Tested by: Matt Jordan
    
    Review: https://reviewboard.asterisk.org/r/1443
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2011-09-20 23:02:25 +00:00
Olle Johansson
73424f128e Merged revisions 336042 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
  
  Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
  
  When using Meetme as a modular call bridge from third party applications, it's handy to make
  it behave like a normal call bridge. When the second to last person exists, the last person
  will be kicked out of the conference when this option is enabled.
  
  (closes issue ASTERISK-18234)
  
  Review: https://reviewboard.asterisk.org/r/1376/
  
  Patch by oej, sponsored by ClearIT, Solna, Sweden
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2011-09-15 12:50:40 +00:00
Jonathan Rose
39fe851e79 Merged revisions 331644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r331644 | jrose | 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) | 9 lines
  
  Merged revisions 331635 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line
    
    Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme
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2011-08-12 18:03:29 +00:00
Jason Parker
1a8069abe2 Merged revisions 331579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r331579 | qwell | 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) | 13 lines
  
  Merged revisions 331578 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines
    
    Use proper values for 64-bit option flags.
    
    Also, reusing bits es no bueno, so change the value of a duplicate.
    
    (issue ASTERISK-18239)
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2011-08-11 21:55:48 +00:00
Kinsey Moore
4ea4b7e1ab Merged revisions 328771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

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  r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
  
  Merged revisions 328770 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
    
    MeetMe requests a PIN twice in some circumstances
    
    If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
    options, MeetMe will ask for the PIN two times: once for creating the
    conference and once for entering the conference.  This behavior was introduced
    in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
    controlling PIN entry for joining a conference.
    
    (closes AST-601)
    Review: https://reviewboard.asterisk.org/r/1305/
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2011-07-19 15:49:55 +00:00
Leif Madsen
a525edea59 Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

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  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
  
  Merged revisions 328209 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
    
    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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2011-07-14 20:28:54 +00:00
Leif Madsen
a5770c43f0 Merged revisions 324176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines
  
  Fix typo in documentation.
  Pointed out by Vlad Povorozniuc
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2011-06-17 18:39:26 +00:00
Richard Mudgett
0096238b52 Merged revisions 320823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
  
  The AMI Newstate event contains different information between v1.4 and v1.8.
  
  The addition of connected line support in v1.8 changes the behavior of the
  channel caller ID somewhat.  The channel caller ID value no longer time
  shares with the connected line ID on outgoing call legs.  The timing of
  some AMI events/responses output the connected line ID as caller ID.
  These party ID's are now separate.
  
  * The ConnectedLineNum and ConnectedLineName headers were added to many
  AMI events/responses if the CallerIDNum/CallerIDName headers were also
  present.
  
  (closes issue #18252)
  Reported by: gje
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1227/
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2011-05-25 17:14:11 +00:00
Richard Mudgett
091fcbce3f Merged revisions 320237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines
  
  Merged revisions 320236 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines
    
    Merged revisions 320235 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines
      
      The meetme CLI command completion leaves conferences mutex locked.
      
      When issuing a meetme kick CLI command and an invalid (non-existent)
      conference number is specified, pressing Tab leaves the conferences mutex
      locked and, therefore, all conferences deadlock.
      
      Add missing unlock.
      
      (closes issue #19336)
      Reported by: zvision
      Patches:
            app_meetme.diff uploaded by zvision (license 798)
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2011-05-20 20:53:30 +00:00
Russell Bryant
6df3b851e3 Merged revisions 317969 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines
  
  Use the right variable to print the time in a debug message.
  
  The original patch also increased some buffer sizes, but that was already
  done in this version.
  
  (closes issue #17034)
  Reported by: sysreq
  Patches:
        asterisk-issue-17034.patch uploaded by sysreq (license 1009)
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2011-05-06 21:49:47 +00:00
Russell Bryant
d05e5281da Merged revisions 317967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines
  
  Fix some more "set but unused" compiler warnings.
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2011-05-06 21:47:05 +00:00
Richard Mudgett
a45d2f29c6 Merged revisions 316831 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines
  
  Wait for leader with Music On Hold allows crosstalk between participants.
  
  Parenthesis in the wrong position.  Regression from issue #14365 when
  expanding conference flags to use 64 bits.
  
  (closes issue #18418)
  Reported by: MrHanMan
  Tested by: rmudgett
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2011-05-04 18:57:02 +00:00
Sean Bright
c596329564 Merged revisions 316476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r316476 | seanbright | 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) | 17 lines
  
  Merged revisions 316475 via svnmerge from 
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    r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines
    
    Honor the C option to MeetMe when L is passed.
    
    This fixes a case that r304773 and friends missed.
    
    (closes issue #17317)
    Reported by: var
    Patches:
          meetme-continue-on-l_16218.diff uploaded by var (license 1227)
    Tested by: seanbright
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2011-05-04 02:39:11 +00:00
Paul Belanger
7c3d14957b Formatting change, remove red blobs
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2011-05-02 15:58:27 +00:00
Olle Johansson
0622568f15 Add explanation of strange flag setup in app_meetme (stolen from Mark's message to asterisk-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-19 08:22:18 +00:00
Jonathan Rose
5af547a619 Minor change to 'L' option for meetme to include some verb statements for the option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-05 13:55:41 +00:00
Brett Bryant
c31d7b21ea Merged revisions 311615 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines
  
  This patch fixes a bug with MeetMe behavior where the 'P' option for always
  prompting for a pin is ignored for the first caller.
  
  (closes issue #18070)
  Reported by: mav3rick
  
  Review: https://reviewboard.asterisk.org/r/1132/
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2011-03-23 21:55:54 +00:00
David Vossel
7902813301 Merged revisions 311497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines
  
  Merged revisions 311496 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines
    
    Fixes memory leak in MeetMe AMI action
  ........
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2011-03-22 15:26:51 +00:00
Jeff Peeler
8f7982f280 Add new manager action MeetmeListRooms.
From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.

(closes issue #17905)
Reported by: rcasas
Patches: 
      app_meetme.c.patch uploaded by rcasas (license 641)

Review: https://reviewboard.asterisk.org/r/874/



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2011-02-09 22:48:02 +00:00
Paul Belanger
3556e4c2d4 Replace ast_log(LOG_DEBUG, ...) with ast_debug()
(closes issue #18556)
Reported by: kkm

Review: https://reviewboard.asterisk.org/r/1071/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 16:55:39 +00:00
David Vossel
c26c190711 Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:22:10 +00:00
Sean Bright
cc2c9442f6 Merged revisions 304777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r304777 | seanbright | 2011-01-29 13:09:37 -0500 (Sat, 29 Jan 2011) | 22 lines
  
  Merged revisions 304776 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) | 15 lines
    
    If we fail to allocate our announcement objects, make sure we don't leak objects.
    
    The majority of this patch was committed already in r304726 and r304729.
    
    (issue #18225)
    Reported by: kenji
    
    (issue #18444)
    Reported by: junky
    
    (closes issue #18343)
    Reported by: kobaz
    Patches:
          meetme-refs.diff uploaded by kobaz (license 834)
  ........
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2011-01-29 18:10:34 +00:00
Sean Bright
ed1ee072b8 Merged revisions 304774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines
  
  Merged revisions 304773 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines
    
    When we pass the S() or L() options to MeetMe, make sure that we honor C as well.
    
    Without this patch, if the user was kicked from the conference via the S() or L()
    mechanism, we would just hang up on them even if we also passed C (continue in
    dialplan when kicked).  With this patch we honor the C flag in those cases.
    
    (closes issue #17317)
    Reported by: var
  ........
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2011-01-29 17:57:01 +00:00
Sean Bright
e229e9f010 Merged revisions 304730 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines
  
  Merged revisions 304729 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines
    
    Make sure that we unref the correct object when ejecting the most recent caller.
    
    Currently, when we kick the last user to enter, we decrement our own reference
    count which results in a crash when we kick another user or when we exit the
    conference ourselves.
    
    This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in
    1.6.2.
    
    (closes issue #18225)
    Reported by: kenji
    Patches:
          issue18225.patch uploaded by seanbright (license 71)
    Tested by: seanbright
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2011-01-29 17:34:22 +00:00
Sean Bright
07b49f3adf Merged revisions 304727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines
  
  Merged revisions 304726 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines
    
    Fix user reference leak in MeetMe.
    
    We were unlinking the user from the conferences user container, but not
    decrementing the reference count of the user as well, resulting in a leak.
    
    (closes issue #18444)
    Reported by: junky
    Tested by: seanbright
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2011-01-29 16:31:17 +00:00
Sean Bright
c5cf436a92 Merged revisions 304683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines
  
  Merged revisions 304659,304682 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines
    
    Don't leak references if we can't create a pseudo channel for mixing in MeetMe.
    
    If there was a problem allocating a pseudo channel when building our meetme, we
    weren't destroying our user container or destroying the mutexes that we created.
  ........
    r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines
    
    Revert part of the previous commit that snuck in.
  ........
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2011-01-28 22:59:27 +00:00
Russell Bryant
092134399c Merged revisions 303549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
  
  Merged revisions 303548 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
    
    Merged revisions 303546 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
      
      Fix channel redirect out of MeetMe() and other issues with channel softhangup.
      
      Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
      working properly.  This issue includes a patch that resolves the issue by
      removing a call to ast_check_hangup() from app_meetme.c.  I left that in my
      patch, as it doesn't need to be there.  However, the rest of the patch fixes
      this problem with or without the change to app_meetme.
      
      The key difference between what happens before and after this patch is the
      effect of the END_OF_Q control frame.  After END_OF_Q is hit in ast_read(),
      ast_read() will return NULL.  With the ast_check_hangup() removed, app_meetme
      sees this which causes it to exit as intended.  Checking ast_check_hangup()
      caused app_meetme to exit earlier in the process, and the target of the
      redirect saw the condition where ast_read() returned NULL.
      
      Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
      solve the issue if another application did the same thing.  There are also
      other edge cases where if an application finishes at the same time that a
      redirect happens, the target of the redirect will think that the channel hung
      up.  So, I made some changes in pbx.c to resolve it at a deeper level.  There
      are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
      abort the hangup process.  My patch extends this to remove the END_OF_Q frame
      from the channel's read queue, making the "abort hangup" more complete.  This
      same technique was used in every place where a softhangup flag was cleared.
      
      (closes issue #18585)
      Reported by: oej
      Tested by: oej, wedhorn, russell
      
      Review: https://reviewboard.asterisk.org/r/1082/
    ........
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2011-01-24 20:57:28 +00:00
Jason Parker
74e0a87776 Merged revisions 301090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines
  
  Merged revisions 301089 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines
    
    Initialize useropts/adminopts in case there is no column in the realtime DB.
    
    (closes issue #18182)
    Reported by: dimas
    Patches: 
          v1-18182.patch uploaded by dimas (license 88)
    Tested by: dimas
  ........
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2011-01-07 20:53:45 +00:00
Russell Bryant
2b056c97cd Merged revisions 297245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines
  
  Merged revisions 297229 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines
    
    Merged revisions 297228 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines
      
      Add "DAHDI" to a couple of app_meetme error messages.
      
      This is in response to some questions on IRC.  To the user, there was nothing
      that made it obvious that this error had anything to do with DAHDI not being
      loaded.
    ........
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2010-12-02 13:20:48 +00:00
Tilghman Lesher
72dc402f1f Merged revisions 296787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) | 2 lines
  
  DOC: Conference number can be omitted; if omitted, all users in a meetme are listed.
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2010-11-30 19:13:54 +00:00
Tilghman Lesher
758a671219 Merged revisions 296467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r296467 | tilghman | 2010-11-27 04:40:22 -0600 (Sat, 27 Nov 2010) | 12 lines
  
  Merged revisions 296466 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) | 5 lines
    
    18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision).
    
    (closes issue #18369)
     Reported by: tnakonz
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2010-11-27 10:41:20 +00:00
Andrew Parisio
935930d8a3 Meetme use voicemail greet for join/leave announce
Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file.  If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.

Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
	meetme_final_patch_v.diff uploaded by parisioa (license 1153)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24 23:46:14 +00:00