Refactored cases where a combination of ast_verbose/options_verbose were
present. Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used. Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2798/
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Merged revisions 397948 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 397958 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r397921 | mmichelson | 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines
Resolve assumptions that bridge snapshots would be non-NULL for transfer stasis events.
Attempting to transfer an unbridged call would result in crashes in either CEL code or
in the conversion to AMI messages.
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r397922 | mmichelson | 2013-08-29 10:42:29 -0500 (Thu, 29 Aug 2013) | 3 lines
Remove extra debug message.
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Merged revisions 397921-397922 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cause code needs to be passed from the disconnecting channel to the
bridge peers if the disconnecting channel dissolves the bridge.
* Made the call to an app_agent_pool agent disconnect with the busy cause
code if the agent does not ack the call in time or hangs up before acking
the call.
(closes issue ASTERISK-22042)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2772/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This essentially makes app_queue usable again. From reviewboard:
* Reporting of transfers and call completion is done by creating stasis
subscriptions and listening for specific events in order to determine
when the call is finished (either via a transfer or hangup).
* Dial end messages have been added where they were previously missing.
* Queue stats are properly being updated again once calls have finished.
* AgentComplete stasis messages and AMI events are now occurring again.
* Mixmonitor starting has been factored into its own function and uses the
Mixmonitor API now instead of using ast_pbx_run()
In addition to the changes in app_queue, there are several supplementary changes as well:
* Queue logging now differentiates between attended and blind transfers. A
note about this is in the CHANGES file.
* Local channel optimization events now report more information. This
includes which of the two local channels involved is the destination of
the optimization, the channel that is replacing the destination local channel,
and an identifier so that begin and end events can be matched to each other.
The end events are now sent whether the optimization was successful or not and
includes an indicator of whether the optimization was successful.
* Changes were made to features and bridging_basic so that additional flags may
be set on a bridge. This is necessary because the queue requires that its
bridge only allows move-swap local channel optimizations into the bridge.
(closes issue ASTERISK-21517)
Reported by Matt Jordan
(closes issue ASTERISK-21943)
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2694
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added an option flags parameter to interval hooks. Interval hooks now
can specify if the callback will affect the media path or not.
* Added an option flags parameter to the bridge action custom callback.
The action callback now can specify if the callback will affect the media
path or not.
* Made the holding bridge technology reexamine the participant idle mode
option whenever the entertainment is restarted.
* Fixed app_agent_pool waiting agents needlessly starting and stopping MOH
every second by specifying the heartbeat interval hook as not affecting
the media path.
* Fixed app_agent_pool agent alert from restarting the MOH after the alert
beep. The agent entertainment is now changed from MOH to silence after
the alert beep.
* Fixed holding bridge technology to defer starting the entertainment. It
was previously a mixture of immediate and deferred.
* Fixed holding bridge technology to immediately stop the entertainment.
It was previously a mixture of immediate and deferred. If the channel
left the bridging system, any deferred stopping was discarded before
taking effect.
* Miscellaneous holding bridge technology rework coding improvements.
Review: https://reviewboard.asterisk.org/r/2761/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Queue members who happen to be in multiple queues at the same time may not
have any wrap up time. This problem occurred due to a code change in Asterisk
11.3.0 that unified device state tracking of Queue members in multiple
Queues (which fixed some other problems, but unfortunately caused this one).
This patch fixes the behavior by having the is_member_available function
check the queue's wrap up time and the time of the member's last call, such
that for a particular queue, the member won't be considered available if their
last call is within the wrap up time.
(closes issue ASTERISK-22189)
Reported by: Tony Lewis
Tested by: Tony Lewis
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Merged revisions 396948 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r382230 added an option to not denoise the MeetMe conference (if a user
had a channel whose format's sample rate changed frequently, for example),
the value added was the maximum allowed value for the constants that define
the options for MeetMe in 1.8. Not so in 11 - unfortunately, the option
CONFFLAG_DONT_DENOISE conflicts with CONFFLAG_INTROUESR_VMREC. This patch
fixes that, and also tweaks one of the way in which the constants was
declared for consistency.
Thanks to Tony Mountifield for pointing out the problem and solution.
(closes issue ASTERISK-22269)
Reported by: Tony Mountifield
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Merged revisions 396944 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The feature_attended_transfer test is failing due to Asterisk not
passing DTMF in the bridges created for internal attended transfers.
This sets the features initialization routine to set this flag by
default and adjusts the basic bridge and confbridge's use of the
bridging system accordingly as per Richard's suggestion instead of
adjusting this individual case. This change allows the necessary DTMF
to pass through the attended transfer bridge and complete the test
successfully.
Review: https://reviewboard.asterisk.org/r/2759/
(closes issue ASTERISK-22222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Holding bridges can allow local channel move/swap optimization to the
bridge. However, we cannot allow it for the BridgeWait holding bridge
because the call will lose the channel roles and dialplan location as a
result.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Changed ast_manager_build_bridge_state_string() to assume an empty
prefix string just like ast_manager_build_channel_state_string().
* Created ast_manager_build_bridge_state_string_prefix() to work just like
ast_manager_build_channel_state_string_prefix().
* Made BridgeMerge AMI event use To/From prefixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This tells
consumers of Stasis that the creation of this channel is an implementation
detail in Asterisk and can be ignored (if they so choose). This
consolidates the conference recorder/announcer flags as well - these flags
had no additional meaning beyond "ignore this channel please".
2. It modifies allocation of a channel in two ways:
(a) If a channel technology can be determined from the name, we set it
directly in the allocation routine. This prevents the initial
publication of the message from going out with a NULL channel technology
where possible. This lets Stasis consumers get the right channel
technology on the first publication.
(b) It reorganizes allocation to make use of the 'finalized' property on the
channel. This was already used to know that a channel had completely
finished its construction in the masquerade routine; now we also use it
to know whether or not the setting of certain channel properties is
occurring during or post construction. The various set routines were
modified accordingly as well.
3. The masquerade event is now dead, Jim. It no longer served any purpose
whatsoever - if you perform a call pickup you'll get a Pickup event;
if you perform an attended transfer you will still get those events; if you
steal a channel to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events.
Review: https://reviewboard.asterisk.org/r/2740
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The rna() routine will raise a Stasis message involving both the caller and the
agent. This doesn't work so well if we already hung up the agent channel, as
the channel doesn't quite exist. Not surprisingly, this will crash. This patch
properly runs the rna subroutine (performing all of the Ring-No-Answer logic)
prior to hanging up the agent channel.
(closes issue ASTERISK-22258)
Reported by: Kiril Valchev
Tested by: Kiril Valchev
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch implements the controls from ARI recordings. The controls
are:
* DELETE /recordings/live/{recordingName} - stop recording and
discard it
* POST /recordings/live/{recordingName}/stop - stop recording
* POST /recordings/live/{recordingName}/pause - pause recording
* POST /recordings/live/{recordingName}/unpause - resume recording
* POST /recordings/live/{recordingName}/mute - mute recording (record
silence to the file)
* POST /recordings/live/{recordingName}/unmute - unmute recording.
Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.
(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability in Queue to raise a hint when a member's paused
state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}',
where {queue_name} and {member_name} are the name of the queue and the name
of the member to subscribe to, respectively.
For example: exten => 8501,hint,Queue:sales_pause_mark.
Members will show as In Use when paused.
Note that the format of the queue pause hint was changed slightly from what
is on the issue to accomodate suggestion on the code review.
Review: https://reviewboard.asterisk.org/r/2254
(closes issue ASTERISK-20842)
Reported by: Philippe Lindheimer
patches:
qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most hook callbacks did not need the bridge parameter. The pointer value
could become invalid if the channel is moved to another bridge while it is
executing.
* Fixed some issues in feature_attended_transfer() as a result.
* Reduce the bridge inhibit count in
attended_transfer_properties_shutdown() after it has restored the bridge
channel hooks.
* Removed basic bridge requirement on feature_blind_transfer(). It does
not require the basic bridge like feature_attended_transfer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed feature limits to not use special members of struct
ast_bridge_features.
* Fixed memory leak in off nominal paths of bridge_builtin_set_limits().
* Fixed off nominal path in ast_bridge_features_limits_construct() freeing
unallocated memory if it was not called by bridge_builtin_set_limits().
* Made bridge_builtin_interval_features.so unloadable.
* Simplified parking's use of its duration interval hook.
* Made BridgeWait S option not depend upon another module being loaded.
(closes issue ASTERISK-22107)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2701/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes arguments for BridgeWait from BridgeWait(role, options) to
BridgeWait(bridge_name, role, options). Now multiple holding bridges may
be created and referenced by this application.
(closes issue ASTERISK-21922)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2642/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Reduced the number of hook containers to just dtmf_hooks,
interval_hooks, and other_hooks. As a result, several functions dealing
with the different hook containers could be combined.
* Extended the generic hook struct for DTMF and interval hooks instead of
using a variant record.
* Merged the special talk detector hook into the other_hooks container.
* Replaced ast_bridge_features_set_talk_detector() with
ast_bridge_talk_detector_hook().
(issue ASTERISK-22107)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It pulls out bridge_channel and puts it into its own translation unit
* It adds public and protected headers for bridging_channel. Protected
functions are appropriate only for the Bridging API and sub-classes of a
bridge.
(issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds new flags to the channel tech properties that flag it as
different types of implementation detail used exclusively to provide a
feature. Examples of channels that would have these flags include the
announcement and recording channels used by confbridge which are the
only two marked as such by this patch.
Review: https://reviewboard.asterisk.org/r/2633/
(closes issue ASTERISK-21873)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds more entertainment options to holding bridges and the
bridge_wait application. Also, holding bridges will now use music on
hold as the default entertainment option instead of none. The
parameters for app_bridgewait have changed to (role, options) from
the previous (options) and the options themselves have changed as
well (entertainment options are now contained in an enumerator, role
specification is handled by the role parameter, etc)
(closes issue ASTERISK-21923)
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/2679/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ast_hangup() is effectively a channel destructor, it should be a
void function.
* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.
* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability to kick all users out of a conference from the
ConfBridge kick CLI command. It is invoked by passing 'all' as the channel
parameter to the CLI command, i.e., "confbridge kick <conf> all".
Note that this patch was modified slightly to conform to trunk.
(closes issue ASTERISK-21827)
Reported by: dorianlogan
patches:
kickall-patch_v2.diff uploaded by dorianlogan (License 6504)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When QUEUE_MEMBER is used and the member specified is not in the queue,
Asterisk provides an ERROR message that indicates that the option specified
is not valid. This patch now properly displays an ERROR message that the
member is not in the queue if an interface is specified.
(closes issue ASTERISK-21980)
Reported by: Avraam David
........
Merged revisions 394345 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk. Worse, support for reloads did not exist at first
and was added later as a bolt-on feature. I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle. Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.
This patch converts various SLA objects to be reference counted objects
using astobj2. This allows reloads to be processed while the system is
in use. The code ensures that the objects will not disappear while one
of the other threads is using them. However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.
Review: https://reviewboard.asterisk.org/r/2581/
........
Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the first step in adding recording support to the
Asterisk REST Interface.
Recordings are stored in /var/spool/recording. Since recordings may be
destructive (overwriting existing files), the API rejects attempts to
escape the recording directory (avoiding issues if someone attempts to
record to ../../lib/sounds/greeting, for example).
(closes issue ASTERISK-21594)
(closes issue ASTERISK-21581)
Review: https://reviewboard.asterisk.org/r/2612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
