This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For some odd reason, loading app_mixmonitor was fine, but res_monitor was not.
This patch fixes a set of issues related to func_periodic_hook exporting the
beep functions that gets res_monitor working again.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add an option to enable a periodic beep to be played into a call if it
is being recorded.  If enabled, it uses the PERIODIC_HOOK() function
internally to play the 'beep' prompt into the call at a specified
interval.  This option is provided for both Monitor() and
MixMonitor().

Review: https://reviewboard.asterisk.org/r/3424/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This app is pretty ancient, so it was never converted to use the
option parsing helper code.  I'd like to add an option to this app
that takes an argument, and that's a pain to do when not using this
helper, so start by doing this conversion.

Review: https://reviewboard.asterisk.org/r/3429/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
    This means CDRs track well with what an actual channel is doing - which
    is useful in transfer scenarios (which were previously difficult to pin
    down). It does, however, mean that CDRs cannot be 'fooled'. Previous
    behavior in Asterisk allowed for CDR applications, channels, and other
    properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
    be what everyone wants, but it is a defined behavior and as such, it is
    predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
    changes have been made to ResetCDR and ForkCDR in particular. Many of the
    options for these two applications no longer made any sense with the new
    framework and the (slightly) more immutable nature of CDRs.

There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.

(closes issue ASTERISK-21196)

Review: https://reviewboard.asterisk.org/r/2486/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When ast_channel_cached_blob_create was merged,
ast_channel_blob_create_from_cache was partially removed in an
unresolved merge conflict. This restores ast_channel_blob_create_from_cache
and refactors usage of ast_channel_cached_blob_create (requires an
ast_channel) to use ast_channel_blob_create_from_cache (requires a
channel uniqueid) instead.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
 * ChanSpyStart/Stop
 * MonitorStart/Stop
 * MusicOnHoldStart/Stop
 * FullyBooted/Reload
 * All Voicemail/MWI related events

In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.

Review: https://reviewboard.asterisk.org/r/2532

(closes issue ASTERISK-21462)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved.  Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.

(Closes issue ASTERISK-19650)
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Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19419)
PseudoReview - seanbright on IRC


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356152 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix potential Solaris crash if Monitor application has a urlbase and no
fname_base option.
........

Merged revisions 355574 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 355575 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I also went ahead and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's
how we handle this stuff these days.

(closes issue ASTERISK-19249)
Reporter: Jamuel Starkey
Patches:
	res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766)
........

Merged revisions 352959 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 352965 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352981 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel.  The channel driver
thread and the PBX thread running dialplan.

* Add lock protection around CDR API calls that access an ast_channel
pointer.

(closes issue ASTERISK-18836)
Reported by: gpluser

Review: https://reviewboard.asterisk.org/r/1628/
........

Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  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
  
  ........
    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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https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines
  
  Merged revisions 328205 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines
    
    Monitor application arguments requirements fixed.
    
    Monitor was requiring options in spite of no individual option on Monitor being required.
    
    Review: https://reviewboard.asterisk.org/r/1320/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328208 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines
  
  When StopMonitor is called, ensure that it will not be restarted by a channel event.
  (closes issue #16590)
   Reported by: kkm
   Patches: 
         resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines
  
  Ensure that monitor recordings are written to the correct location (again)
  
  This is an extension to 248860. As such the dialplan test has been extended:
  
  ; non absolute path, not combined
  exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
  exten => 5040, n, dial(sip/5001)
  ; absolute path, not combined
  exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2)
  exten => 5041, n, dial(sip/5001)
  ; no path, not combined
  exten => 5042, 1, monitor(wav,monitor_test3)
  exten => 5042, n, dial(sip/5001)
  ; combined: changemonitor from non absolute to no path (leaves tmp/jeff)
  exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
  exten => 5043, n, changemonitor(monitor_test5)
  exten => 5043, n, dial(sip/5001)
  ; combined: changemonitor from no path to non absolute path
  exten => 5044, 1, monitor(wav,monitor_test6,m)
  exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before
  exten => 5044, n, dial(sip/5001)
  ; non absolute path, combined
  exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m)
  exten => 5045, n, dial(sip/5001)
  ; absolute path, combined 
  exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m)
  exten => 5046, n, dial(sip/5001)
  ; no path, combined
  exten => 5047, 1, monitor(wav,monitor_test10,m)
  exten => 5047, n, dial(sip/5001)
  ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
  exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m)
  exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12)
  exten => 5048, n, dial(sip/5001)
  ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
  exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m)
  exten => 5049, n, changemonitor(tmp/jeff/monitor_test14)
  exten => 5049, n, dial(sip/5001)
  ; combined: changemonitor from no path to absolute
  exten => 5050, 1, monitor(wav,monitor_test15,m)
  exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16)
  exten => 5050, n, dial(sip/5001)
  ; combined: changemonitor from absolute to no path (leaves /tmp/jeff)
  exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m)
  exten => 5051, n, changemonitor(monitor_test18)
  exten => 5051, n, dial(sip/5001)
  ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff)
  exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
  exten => 5052, n, changemonitor(monitor_test20)
  exten => 5052, n, dial(sip/5001)
  ; not combined: changemonitor from no path to non absolute
  exten => 5053, 1, monitor(wav,monitor_test21)
  exten => 5053, n, changemonitor(tmp/jeff/monitor_test22)
  exten => 5053, n, dial(sip/5001)
  ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
  exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23)
  exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24)
  exten => 5054, n, dial(sip/5001)
  ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
  exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24)
  exten => 5055, n, changemonitor(tmp/jeff/monitor_test25)
  exten => 5055, n, dial(sip/5001)
  ; not combined: changemonitor from no path to absolute
  exten => 5056, 1, monitor(wav,monitor_test26)
  exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27)
  exten => 5056, n, dial(sip/5001)
  ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff)
  exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28)
  exten => 5057, n, changemonitor(monitor_test29)
  exten => 5057, n, dial(sip/5001)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254277 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines
  
  Ensure that monitor recordings are written to the correct location (again)
  
  This is an extension to 248757. As such the dialplan test has been extended:
  
  exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
  exten => 5040, n, dial(sip/5001)
  exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
  exten => 5041, n, dial(sip/5001)
  exten => 5042, 1, monitor(wav,monitor_test3,b)
  exten => 5042, n, dial(sip/5001)
  exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m)
  exten => 5043, n, changemonitor(monitor_test4)
  exten => 5043, n, dial(sip/5001)
  exten => 5044, 1, monitor(wav,monitor_test4,m)
  exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning
  exten => 5044, n, dial(sip/5001)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines
  
  Ensure that monitor recordings are written to the correct location.
  
  Recordings should be placed in the monitor directory when a non-absolute path
  is used.
  
  Exact dialplan used for testing:
  exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
  exten => 5040, n, dial(sip/5001)
  exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
  exten => 5041, n, dial(sip/5001)
  exten => 5042, 1, monitor(wav,monitor_test3,b)
  exten => 5042, n, dial(sip/5001)
  
  ABE-2101
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines
  
  Change Monitor to not assume file to write to does not contain pathing.
  
  227944 changed the fname_base argument to always append the configured monitor
  path. This change was necessary to properly compare files for uniqueness. 
  If a full path is given though, nothing needs to be appended and that is
  handled correctly now.
  
  (closes issue #16377)
  (closes issue #16376)
  Reported by: bcnit
  Patches:
        res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235941 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(Closes AST-33)
Reviewboard:	https://reviewboard.asterisk.org/r/368/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines
  
  Fix incorrect filename comparsion after monitor file change
  
  The logic to detect if a requested file is indeed a different file from the
  current file was incorrect. The main issue being confusion of the use of
  filename_base which was previously set without pathing information and then
  compared to another full path. Robust file comparison logic has been added
  to properly check if two files are the same even if symlinks are used.
  
  (closes issue #15313)
  Reported by: caspy
  Patches: 
        20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325)
        but mostly tilghman's work
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

a result of the const-ify the world patch.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines
  
  Add a similar dependency on SMDI for voicemail as already exists for ADSI.
  (closes issue #14846)
   Reported by: pj
   Patches: 
         20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14)
         20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14)
         20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195370 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There is a lot that could be said about this, but the patch is a big 
improvement for performance, stability, code maintainability, 
and ease of future code development.

The channel list is no longer an unsorted linked list.  The main container 
for channels is an astobj2 hash table.  All of the code related to searching 
for channels or iterating active channels has been rewritten.  Let n be 
the number of active channels.  Iterating the channel list has gone from 
O(n^2) to O(n).  Searching for a channel by name went from O(n) to O(1).  
Searching for a channel by extension is still O(n), but uses a new method 
for doing so, which is more efficient.

The ast_channel object is now a reference counted object.  The benefits 
here are plentiful.  Some benefits directly related to issues in the 
previous code include:

1) When threads other than the channel thread owning a channel wanted 
   access to a channel, it had to hold the lock on it to ensure that it didn't 
   go away.  This is no longer a requirement.  Holding a reference is 
   sufficient.

2) There are places that now require less dealing with channel locks.

3) There are places where channel locks are held for much shorter periods 
   of time.

4) There are places where dealing with more than one channel at a time becomes 
   _MUCH_ easier.  ChanSpy is a great example of this.  Writing code in the 
   future that deals with multiple channels will be much easier.

Some additional information regarding channel locking and reference count 
handling can be found in channel.h, where a new section has been added that 
discusses some of the rules associated with it.

Mark Michelson also assisted with the development of this patch.  He did the 
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it 
much easier to deal with holding on to a channel pointer for an extended period 
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.

Thanks to David Vossel for his assistance with this branch, as well.  David 
did the conversion of the DAHDIScan application by making it become a wrapper 
for ChanSpy internally.

The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.

Review: http://reviewboard.digium.com/r/203/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
  
  Fix a memory leak in res_monitor.c
  
  The only way that this leak would occur is if Monitor were started
  using the Manager interface and no File: header were given. Discovered
  while reviewing the ast_channel_ao2 review request.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14065)
Reported by: kshumard
Patches:
      res_monitor.documentation.patch.txt uploaded by kshumard (license 92)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163612 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148070 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12034)
Reported by: kowalma


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan 2008) | 10 lines

1. Prevent the addition of an extra '/' to the beginning of an absolute pathname.
2. If ast_monitor_change_fname is called and the new filename is the same as the old, then exit early and don't set the
   filename_changed field in the monitor structure. Setting it in this case was causing ast_monitor_stop to erroneously
   delete them.

(closes issue #11741)
Reported by: garlew
Tested by: putnopvut


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101532 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec 2007) | 6 lines

Sox versions 13.0.0 and newer do not have "soxmix" and instead use sox -m. res_monitor
needs to use this if the user does not have soxmix.

(closes issue #11589, reported by amessina, patch inspired by amessina but with a flourish from me)


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94123 65c4cc65-6c06-0410-ace0-fbb531ad65f3

modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #11505)
Reported by: Mike Anikienko


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92365 65c4cc65-6c06-0410-ace0-fbb531ad65f3