Commit Graph

139 Commits

Author SHA1 Message Date
Richard Mudgett e6738b79b3 Complete deprecating legacy modules.
The menuselect comment was updated to deprecate these modules but the
AST_MODULE_INFO block at the end of file was missed.

ASTERISK-27671

Change-Id: I63070b5c4d4f08af010c6034acd4793c1bcef839
2018-03-08 13:53:09 -06:00
Corey Farrell 9f74afbdcf Deprecate legacy modules.
* app_fax (replaced by res_fax).
* res_config_sqlite (replaced by res_config_sqlite3).
* res_monitor (replaced by app_mixmonitor).

This is related to ASTERISK~23657 but does not resolve that ticket.
Resolving that ticket would require complete removal of res_monitor.

ASTERISK-27671 #close

Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49
2018-02-13 13:56:03 -05:00
Jenkins2 a231e1d155 Merge "Remove redundant module checks and references." 2018-01-29 12:39:34 -06:00
Corey Farrell 527cf5a570 Remove redundant module checks and references.
This removes references that are no longer needed due to automatic
references created by module dependencies.

In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.

Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
2018-01-24 13:37:29 -05:00
Corey Farrell 679fa5fb34 Add missing OPTIONAL_API and ARI dependences.
I've audited all modules that include any header which includes
asterisk/optional_api.h.  All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.

In addition ARI dependency declarations have been reworked.  Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.

Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606
2018-01-22 12:16:58 -05:00
Sean Bright fd0ca1c3f9 Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
2017-12-22 09:23:22 -05:00
Joshua Colp 68b6ebd836 Merge "Prevent unload of modules which implement an Optional API." 2017-11-06 10:11:51 -06:00
Corey Farrell 606ae3484a Add missing menuselect dependencies.
This adds menuselect dependencies for modules that use symbols of other
modules.

ASTERISK-27390

Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385
2017-11-02 02:57:52 -04:00
Corey Farrell 79f111e1f3 Prevent unload of modules which implement an Optional API.
Once an Optional API module is loaded it should stay loaded.  Unloading
an optional API module runs the risk of a crash if something else is
using it.  This patch causes all optional API providers to tell the
module loader not to unload except at shutdown.

ASTERISK-27389

Change-Id: Ia07786fe655681aec49cc8d3d96e06483b11f5e6
2017-11-01 20:46:11 -04:00
Corey Farrell 1bf3dfffd7 AST-2017-006: Fix app_minivm application MinivmNotify command injection
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received.  The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.

* Add ast_safe_execvp() function.  This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding.  This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.

* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.

* Document code injection potential from untrusted data sources for other
shell commands that are under user control.

ASTERISK-27103

Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
2017-08-30 18:43:38 +00:00
Corey Farrell 58d032112b Fix compiler warnings on Fedora 26 / GCC 7.
GCC 7 has added capability to produce warnings, this fixes most of those
warnings.  The specific warnings are disabled in a few places:

* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().

ASTERISK-27156 #close

Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
2017-08-01 15:42:38 -06:00
Corey Farrell a6e5bae3ef Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
2016-10-27 09:53:55 -04:00
Rodrigo Ramírez Norambuena eec010829a AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.
Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
2015-05-13 16:34:23 -05:00
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00
Matthew Jordan 49aa87e17c res/res_monitor: Reset in/out sample counts on Monitor start
When repeatedly starting/stopping a Monitor on a channel, the accumulated
in/out sample counts are never reset to 0. This can cause inadvertent jumps
in the recordings, as the code in the channel core will determine incorrectly
that a jump in the recorded file position should occur. Setting the sample
counts to 0 simply reflects the initial state a Monitor should be in when it
is started, as this is the initial count that would be on the channels at that
time.

ASTERISK-24573 #close
Reported by: Nuno Borges
patches:
  24573.patch uploaded by Nuno Borges (License 6116)
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2014-12-06 18:16:49 +00:00
Mark Michelson dcf1ad14da Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/3802



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25 16:47:17 +00:00
Kinsey Moore abd3e4040b Allow Asterisk to compile under GCC 4.10
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.
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2014-05-09 22:49:26 +00:00
Russell Bryant 4b9b4790d9 Fix error loading res_monitor.
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
2014-04-23 15:02:39 +00:00
Russell Bryant 5b7a769fd8 (mix)monitor: Add options to enable a periodic beep
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
2014-04-15 23:21:19 +00:00
Russell Bryant af95d8c1d2 monitor: use app options parsing helper code
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
2014-04-11 01:12:54 +00:00
Matthew Jordan 6258bbe7bd Update Asterisk's CDRs for the new bridging framework
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
2013-06-17 03:00:38 +00:00
Kinsey Moore 6851801a5e Resolve a merge conflict
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
2013-05-29 02:26:17 +00:00
Matthew Jordan 06be8463b6 Migrate a large number of AMI events over to Stasis-Core
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
2013-05-24 20:44:07 +00:00
Kinsey Moore dd81b047db Resolve FORWARD_NULL static analysis warnings
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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2012-05-10 20:56:09 +00:00
Terry Wilson c25a442dfb Fix some opaquification-related compiler warnings
(closes issue ASTERISK-19419)
PseudoReview - seanbright on IRC


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-21 20:17:52 +00:00
Terry Wilson 57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Richard Mudgett 7879cccafd Fix AMI Monitor action without File header converting channel name into filename.
* Fix potential Solaris crash if Monitor application has a urlbase and no
fname_base option.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-16 18:39:46 +00:00
Jonathan Rose 8a401484da Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session.
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-27 19:26:53 +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/


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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
Richard Mudgett b05d4603c4 Fix crash during CDR update.
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/
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2011-12-16 21:10:19 +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
Jonathan Rose 8dc71df9d0 Merged revisions 328207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

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  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
  
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    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/
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2011-07-14 19:56:19 +00:00
Tilghman Lesher b4e18d5660 Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-20 19:35:02 +00:00
Tilghman Lesher 56a6994310 Merged revisions 258775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  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)
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2010-04-25 18:12:14 +00:00
Jeff Peeler 5990fe07b8 Merged revisions 254235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  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
2010-03-24 17:15:05 +00:00
Jeff Peeler 406bb18127 Merged revisions 248860 via svnmerge from
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
2010-02-25 23:09:54 +00:00
Jeff Peeler d64987f8ad Merged revisions 248757 via svnmerge from
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
2010-02-25 18:37:56 +00:00
Jeff Peeler f95ee393f2 Merged revisions 235940 via svnmerge from
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
2009-12-21 19:54:20 +00:00
Tilghman Lesher 5e2aa190fe Display a list of channel variables in each channel-oriented event.
(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
2009-11-13 20:42:03 +00:00
Jeff Peeler b6290e0e93 Merged revisions 227944 via svnmerge from
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
2009-11-04 23:50:59 +00:00
Kevin P. Fleming 6c5987811c Redesigned 'optional API' support.
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
2009-06-15 16:07:23 +00:00
Eliel C. Sardanons fb73ee6187 Moved more static documentation to the new AstXML form.
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
2009-06-01 19:37:30 +00:00
Sean Bright fcda626f3c Fix build under dev mode and remove some casts that are no longer necessary as
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
2009-05-22 16:10:33 +00:00
Kevin P. Fleming e6b2e9a750 Const-ify the world (or at least a good part of it)
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
2009-05-21 21:13:09 +00:00
Tilghman Lesher bdcafc1ab4 Recorded merge of revisions 195366 via svnmerge from
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
2009-05-18 20:52:33 +00:00
Russell Bryant cba19c8a67 Convert the ast_channel data structure over to the astobj2 framework.
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
2009-04-24 14:04:26 +00:00
Mark Michelson 85cbd1fd46 Merged revisions 183700 via svnmerge from
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
2009-03-23 18:58:03 +00:00
Michiel van Baak 39e47e88e9 Document default Monitor file location.
(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
2008-12-12 17:22:47 +00:00
Tilghman Lesher 8b14e5f493 Reverting format addition for now
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 21:47:02 +00:00