Commit graph

659 commits

Author SHA1 Message Date
Matthew Jordan
b172d369c4 res_pjsip: Add PJSIP CLI commands
Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)

Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.

New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.

(issue ASTERISK-22610)
patches:
  pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
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2013-12-20 21:32:13 +00:00
Mark Michelson
1b91ee6c4b Fix a deadlock that occurred due to a conflict of masquerades.
For the explanation, here is a copy-paste of the review board explanation:

Initially, it was discovered that performing an attended transfer of a
multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread
started a masquerade and reached the point where it was calling the fixup()
callback on the "original" channel. For chan_pjsip, this involves pushing a
synchronous task to the session's serializer. The problem was that a task ahead
of the fixup task was also attempting to perform a channel masquerade. However,
since masquerades are designed in a way to only allow for one to occur at a
time, the task ahead of the fixup could not continue until the masquerade
already in progress had completed. And of course, the masquerade in progress
could not complete until the task ahead of the fixup task had completed.
Deadlock.

The initial fix was to change the fixup task to be asynchronous. While this
prevented the deadlock from occurring, it had the frightful side effect of
potentially allowing for tasks in the session's serializer to operate on a
zombie channel.

Taking a step back from this particular deadlock, it became clear that the
problem was not really this one particular issue but that masquerades
themselves needed to be addressed. A PJSIP attended transfer operation calls
ast_channel_move(), which attempts to both set up and execute a masquerade. The
problem was that after it had set up the masquerade, the PBX thread had swooped
in and tried to actually perform the masquerade. Looking at changes that had
been made to Asterisk 12, it became clear that there never is any time now that
anyone ever wants to set up a masquerade and allow for the channel thread to
actually perform the masquerade. Everyone always is calling ast_channel_move(),
performs the masquerade itself before returning.

In this patch, I have removed all blocks of code from channel.c that will
attempt to perform a masquerade if ast_channel_masq() returns true. Now, there
is no distinction between setting up a masquerade and performing the
masquerade. It is one operation. The only remaining checks for
ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not
want to interrupt a masquerade by hanging up the channel. Instead, now
ast_hangup() will wait for a masquerade to complete before moving forward with
its operation.

The ast_channel_move() function has been modified to basically in-line the
logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has
been killed off for real. ast_channel_move() now has a lock associated with it
that is used to prevent any simultaneous moves from occurring at once. This
means there is no need to make sure that ast_channel_masq() or
ast_channel_masqr() are already set on a channel when ast_channel_move() is
called. It also means the channel container lock is not pulling double duty by
both keeping the container locked and preventing multiple masquerades from
occurring simultaneously.

The ast_do_masquerade() function has been renamed to do_channel_masquerade()
and is now internal to channel.c. The function now takes explicit arguments of
which channels are involved in the masquerade instead of a single channel.
While it probably is possible to do some further refactoring of this method, I
feel that I would be treading dangerously. Instead, all I did was change some
comments that no longer are true after this changeset.

The other more minor change introduced in this patch is to res_pjsip.c to make
ast_sip_push_task_synchronous() run the task in-place if we are already a SIP
servant thread. This is related to this patch because even when we isolate the
channel masquerade to only running in the SIP servant thread, we would still
deadlock when the fixup() callback is reached since we would essentially be
waiting forever for ourselves to finish before actually running the fixup. This
makes it so the fixup is run without having to push a task into a serializer at
all.

(closes issue ASTERISK-22936)
Reported by Jonathan Rose

Review: https://reviewboard.asterisk.org/r/3069
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2013-12-19 17:45:21 +00:00
Joshua Colp
eb235ad05f channel: Add a missing ast_channel_unlock when allocating a Surrogate channel.
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Merged revisions 404332 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-19 12:18:52 +00:00
Kevin Harwell
28c0cb28d0 channel locking: Add locking for channel snapshot creation
Original commit message by mmichelson (asterisk 12 r403311):

"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."

The above was initially committed and then reverted at r403398.  The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed.  Fixed by unreffing the channels.

Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel.  Fixed by
unlocking "other->chan"

(closes issue ASTERISK-22709)
Reported by: John Bigelow
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Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-18 20:33:37 +00:00
Joshua Colp
e2630fcd51 channels: Return allocated channels locked.
This change makes ast_channel_alloc return allocated channels
locked. By doing so no other thread can acquire, lock, and manipulate
the channel before it is completely set up.

(closes issue AST-1256)

Review: https://reviewboard.asterisk.org/r/3067/
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2013-12-18 19:28:05 +00:00
Rusty Newton
f7c60b8fb6 Several components: fixing Typos in comments and code, "avaliable" instead of "available"
(issue ASTERISK-23021)
(closes issue ASTERISK-23021)
Reported by: Jeremy Lainé
Tested by: Rusty Newton
Patches:
   available.patch uploaded by Jeremy Lainé (license 6561)
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2013-12-17 23:38:02 +00:00
Kevin Harwell
84e1790beb bridge_native_rtp: Deadlock during 4-way conference creation
The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore).  A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback.  Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.

(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
     lock_inversion.diff uploaded by kmoore (license 6273)
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2013-12-13 18:33:25 +00:00
David M. Lee
1212906351 Reverting r403311. It's causing ARI tests to hang.
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2013-12-05 22:10:20 +00:00
Mark Michelson
8e8b329e14 Add channel locking for channel snapshot creation.
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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2013-12-03 17:07:29 +00:00
Mark Michelson
94f19c8218 Switch to a scoped lock to avoid missing unlocks in failure returns.
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2013-11-12 19:38:03 +00:00
Mark Michelson
c0bc3f6b4c Move a NULL check to a place that makes more sense.
Two variables were being checked for NULLity immediately
after being declared NULL. I moved the NULL check until
after the variables are allocated.

This allows for the "channelvars" option in manager.conf
to work as intended again.
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2013-11-12 19:08:14 +00:00
Matthew Jordan
076b29dd5b Remove some spammy debug messages; improve clarity of others
Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:

* channel.c: When copying variables from a parent channel to a child channel,
  specify the channels involved. Do not log anything for a variable that is not
  inherited; the fact that it doesn't have an _ or __ already signifies that it
  won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
  to use these debug messages, and for each format that is registered (on
  startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
  For short tests in the Asterisk Test Suite, this should make finding the
  actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
  Often, description elements - which are not required - are not provided.
  This debug message adds no additional value, as it is not indicative of an
  error or helpful in debugging which element did not contain a 'blah' element
  as a child. If an element is supposed to contain a child element, then that
  XML tree should have failed validation in the first place.

Review: https://reviewboard.asterisk.org/r/2966/
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2013-10-29 12:57:35 +00:00
Richard Mudgett
057d105c5a Add channel lock protection around translation path setup.
Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge.  With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.

* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.

* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper().  The call to
ast_translator_best_choice() got them backwards.

* Updated some callers of ast_channel_make_compatible() and the function
documentation.  There is actually a difference between the two channels
passed in.

* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible().  The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.

(closes issue ASTERISK-22542)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2915/
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2013-10-18 16:59:09 +00:00
Walter Doekes
f33e0776ec Properly copy/remove the device state cache flag over a masquerade.
In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells
the devstate system to not cache states for non-real devices. However,
when optimizing away channels (ast_do_masquerade), that flag wasn't
copied.

In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.

(closes issue ASTERISK-22718)
Reported by: Walter Doekes

Review: https://reviewboard.asterisk.org/r/2925/
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2013-10-18 14:50:27 +00:00
Mark Michelson
ee21eee7e0 Cache string values of formats on ast_format_cap() to save processing.
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.

Review: https://reviewboard.asterisk.org/r/2879
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2013-10-03 14:58:16 +00:00
Joshua Colp
c1235f2639 Reduce channel snapshot creation and publishing by up to 50%.
This change introduces the ability to stage channel snapshot
creation and publishing by suppressing the implicit creation
and publishing that some functions have. Once all operations
are executed the staging is marked as done and a single snapshot
is created and published.

Review: https://reviewboard.asterisk.org/r/2889/
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2013-10-02 16:23:34 +00:00
David M. Lee
2de42c2a25 Multiple revisions 399887,400138,400178,400180-400181
........
  r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
  
  Minor performance bump by not allocate manager variable struct if we don't need it
........
  r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
  
  Stasis performance improvements
  
  This patch addresses several performance problems that were found in
  the initial performance testing of Asterisk 12.
  
  The Stasis dispatch object was allocated as an AO2 object, even though
  it has a very confined lifecycle. This was replaced with a straight
  ast_malloc().
  
  The Stasis message router was spending an inordinate amount of time
  searching hash tables. In this case, most of our routers had 6 or
  fewer routes in them to begin with. This was replaced with an array
  that's searched linearly for the route.
  
  We more heavily rely on AO2 objects in Asterisk 12, and the memset()
  in ao2_ref() actually became noticeable on the profile. This was
  #ifdef'ed to only run when AO2_DEBUG was enabled.
  
  After being misled by an erroneous comment in taskprocessor.c during
  profiling, the wrong comment was removed.
  
  Review: https://reviewboard.asterisk.org/r/2873/
........
  r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
  
  Taskprocessor optimization; switch Stasis to use taskprocessors
  
  This patch optimizes taskprocessor to use a semaphore for signaling,
  which the OS can do a better job at managing contention and waiting
  that we can with a mutex and condition.
  
  The taskprocessor execution was also slightly optimized to reduce the
  number of locks taken.
  
  The only observable difference in the taskprocessor implementation is
  that when the final reference to the taskprocessor goes away, it will
  execute all tasks to completion instead of discarding the unexecuted
  tasks.
  
  For systems where unnamed semaphores are not supported, a really
  simple semaphore implementation is provided. (Which gives identical
  performance as the original taskprocessor implementation).
  
  The way we ended up implementing Stasis caused the threadpool to be a
  burden instead of a boost to performance. This was switched to just
  use taskprocessors directly for subscriptions.
  
  Review: https://reviewboard.asterisk.org/r/2881/
........
  r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Optimize how Stasis forwards are dispatched
  
  This patch optimizes how forwards are dispatched in Stasis.
  
  Originally, forwards were dispatched as subscriptions that are invoked
  on the publishing thread. This did not account for the vast number of
  forwards we would end up having in the system, and the amount of work it
  would take to walk though the forward subscriptions.
  
  This patch modifies Stasis so that rather than walking the tree of
  forwards on every dispatch, when forwards and subscriptions are changed,
  the subscriber list for every topic in the tree is changed.
  
  This has a couple of benefits. First, this reduces the workload of
  dispatching messages. It also reduces contention when dispatching to
  different topics that happen to forward to the same aggregation topic
  (as happens with all of the channel, bridge and endpoint topics).
  
  Since forwards are no longer subscriptions, the bulk of this patch is
  simply changing stasis_subscription objects to stasis_forward objects
  (which, admittedly, I should have done in the first place.)
  
  Since this required me to yet again put in a growing array, I finally
  abstracted that out into a set of ast_vector macros in
  asterisk/vector.h.
  
  Review: https://reviewboard.asterisk.org/r/2883/
........
  r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Remove dispatch object allocation from Stasis publishing
  
  While looking for areas for performance improvement, I realized that an
  unused feature in Stasis was negatively impacting performance.
  
  When a message is sent to a subscriber, a dispatch object is allocated
  for the dispatch, containing the topic the message was published to, the
  subscriber the message is being sent to, and the message itself.
  
  The topic is actually unused by any subscriber in Asterisk today. And
  the subscriber is associated with the taskprocessor the message is being
  dispatched to.
  
  First, this patch removes the unused topic parameter from Stasis
  subscription callbacks.
  
  Second, this patch introduces the concept of taskprocessor local data,
  data that may be set on a taskprocessor and provided along with the data
  pointer when a task is pushed using the ast_taskprocessor_push_local()
  call. This allows the task to have both data specific to that
  taskprocessor, in addition to data specific to that invocation.
  
  With those two changes, the dispatch object can be removed completely,
  and the message is simply refcounted and sent directly to the
  taskprocessor.
  
  Review: https://reviewboard.asterisk.org/r/2884/
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2013-09-30 18:55:27 +00:00
Richard Mudgett
2a371cd80b Restore Dial, Queue, and FollowMe 'I' option support.
The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.

* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.

* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.

* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.

* Made all callers of ast_bridge_impart() check the return value.  It is
important.  As a precaution, I also made the compiler complain now if it
is not checked.

* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.

An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.

(closes issue ASTERISK-22072)
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/2845/
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2013-09-13 22:19:23 +00:00
Richard Mudgett
6ebfac8e70 Handle DTMF and hold wrapup when a channel leaves the bridging system.
DTMF start/end and hold/unhold events have state because a DTMF begin
event and hold event must be ended by something.

The following cases need to be handled when a channel is moved around in
the system.

* When a channel leaves a bridge it may owe a DTMF end event to the
bridge.

* When a channel leaves a bridge it may owe an UNHOLD event to the bridge.
(This case is explicitly ignored because things like transfers need
explicit control over this.)

* When a channel leaves the bridging system it may need to simulate a DTMF
end event to the channel.

* When a channel leaves the bridging system it may need to simulate an
UNHOLD event to the channel.

The patch also fixes the following:
* Fixes playing a file and restarting MOH using the latest MOH class used.

(closes issue ASTERISK-22043)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2791/


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2013-08-23 18:33:36 +00:00
Matthew Jordan
4d348e853c Add pass through support for Opus and VP8; Opus format attribute negotiation
This patch adds pass through support for Opus and VP8. That includes:

* Format attribute negotiation for Opus. Note that unlike some other codecs,
  the draft RFC specifies having spaces delimiting the attributes in addition
  to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
  chan_sip, so a small tweak was also included in this patch for that.

* A format attribute negotiation module for Opus, res_format_attr_opus

* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
  than FIR, this really is specific to VP8 at this time.

Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.

Review: https://reviewboard.asterisk.org/r/2723/

(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
  asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23 15:42:27 +00:00
Richard Mudgett
c25c093c67 Minor tweaks with ast_moh_start() callers.
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2013-08-22 23:15:14 +00:00
Kinsey Moore
7b032c1adb Add SayAlphaCase and similar functionality for AGI
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
2013-08-22 22:33:48 +00:00
Richard Mudgett
62c2b80487 Remove unsupported channel technology callbacks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15 00:16:39 +00:00
Matthew Jordan
33e7b76d1d Hide the Surrogate channels from external consumers; kill Masquerade events
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
2013-08-08 14:13:05 +00:00
David M. Lee
c790848794 ARI: Add recording controls
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
2013-08-06 14:44:45 +00:00
David M. Lee
357b275239 Fix res_ari_asterisk load issue
The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.

This patch renames the variables, adding the ast_ prefix so they will
be exported.

Review: https://reviewboard.asterisk.org/r/2737


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-05 14:35:00 +00:00
Mark Michelson
f8622e7c5c Get rid of ast_bridged_channel() and the bridged_channel field on ast_channels.
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
2013-08-02 14:05:07 +00:00
Matthew Jordan
cafc115896 A great big renaming patch
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
2013-07-25 04:06:32 +00:00
Richard Mudgett
70fbe9dc14 Add missing end-of-file line terminators.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-24 16:01:20 +00:00
Matthew Jordan
d91dc6d1a8 Perform the initial renaming of the Bridging API
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
2013-07-24 15:38:18 +00:00
Matthew Jordan
ff2f5eaa23 Kill the zombies
In previous versions of Asterisk, the zombies roamed freely,
unchecked and uncontrolled. They ravaged Asterisk systems with
their biting and their nashing and their pointy teeth.

Sometimes, you couldn't even hang them up.

Now, zombies are rare. They still *technically* exist in certain
places, but they are controlled. Kind of like a zombie zoo: you can
see them, but you can't touch them, and they can't touch you.
Bring your kids!

Because zombies are now population controlled with a very short lifespan,
there's no reason to rename the channels to '%s<ZOMBIE>'. The channels
are guaranteed to die off quickly; the rename really is just confusing
at this point.

This patch finally removes the renaming. On the plus side: this made
my life easier in CDRs during call pickup and attended transfers to
an Asterisk application. It will make other folks lives easier as well!

Review: https://reviewboard.astierks.org/r/2690/

(closes issue ASTERISK-21699)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23 14:49:53 +00:00
Kinsey Moore
684c83b29b Add transfer support to CEL
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
2013-07-20 13:10:22 +00:00
Richard Mudgett
643fb1ed14 Minor optimizations.
* Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty()
NULL tolerant.

* Made ast_audiohook_detach_list() return void since it is a destructor.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-19 23:30:10 +00:00
Richard Mudgett
2838683742 Extract a repeated test into ast_channel_has_audio_frame_or_monitor().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-19 22:47:10 +00:00
Jason Parker
c1a7567d24 ARI: Add support for suppressing media streams.
Also convert res_mutestream to use the core feature behind this.

(closes issue ASTERISK-21618)

Review: https://reviewboard.asterisk.org/r/2652/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-18 16:03:12 +00:00
Richard Mudgett
40ce5e0d18 Change ast_hangup() to return void and be NULL safe.
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
2013-07-17 22:30:28 +00:00
Richard Mudgett
6ba25dd3f2 Remove some dead code dealing with old bridging method.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-16 18:49:26 +00:00
Joshua Colp
238a54fa15 Add support to the bridging core for performing COLP updates when channels join a 2 party bridge.
(closes issue ASTERISK-21829)

Review: https://reviewboard.asterisk.org/r/2636/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-12 21:42:53 +00:00
Joshua Colp
7c044acbd9 Refactor operations to access the stasis cache instead of objects directly when retrieving information.
(closes issue ASTERISK-21883)

Review: https://reviewboard.asterisk.org/r/2645/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08 19:19:55 +00:00
Matthew Jordan
70a46e2ee5 In a channel destructor dispose of items that raise Stasis message properly
This patch reorders certain actions that may raise Stasis messages in the
channel destructor such that they occur before the Stasis cache is cleared.
Once the Stasis cache is cleared, its rather a bad idea to be trying to
publish information about a channel.

(closes issue ASTERISK-22001)
Reported by: Jonathan Rose



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-07 21:29:40 +00:00
Matthew Jordan
b193c2873d Handle hangup logic in the Stasis message bus and consumers of Stasis messages
This patch does the following:
* It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a
  channel is executing dialplan hangup logic, i.e., the 'h' extension or a
  hangup handler. Stasis messages now also convey the soft hangup flag so
  consumers of the messages can know when a channel is executing said
  hangup logic.
* It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is
  well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs,
  and other consumers of Stasis have been updated to look for this flag to
  know when the channel should by lying six feet under.
* The CDR engine has been updated to better handle a channel entering and
  leaving a bridge. Previously, a new CDR was automatically created when a
  channel left a bridge and put into the 'Pending' state; however, this
  way of handling CDRs made it difficult for the 'endbeforehexten' logic to
  work correctly - there was always a new CDR waiting in the hangup logic
  and, even if 'ended', wouldn't be the CDR people wanted to inspect in the
  hangup routine. This patch completely removes the Pending state and instead
  defers creation of the new CDR until it gets a new message that requires
  a new CDR.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-07 20:34:38 +00:00
Jonathan Rose
93ed5ef0ff res_parking: Replace Parker snapshots with ParkerDialString
This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.

(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-04 18:46:56 +00:00
David M. Lee
a75fd32212 ARI - channel recording support
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
2013-07-03 17:58:45 +00:00
Richard Mudgett
291711f85f chan_vpb: Fix compile error and __ast_channel_alloc() prototype const inconsistency.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 18:37:27 +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
4f84e48028 Refactor CEL channel events on top of Stasis-Core
This uses the channel state change events from Stasis-Core to determine
when channel-related CEL events should be raised. Those refactored in
this patch are:
* AST_CEL_CHANNEL_START
* AST_CEL_ANSWER
* AST_CEL_APP_START
* AST_CEL_APP_END
* AST_CEL_HANGUP
* AST_CEL_CHANNEL_END

Retirement of Linked IDs is also refactored.

CEL configuration has been refactored to use the config framework.

Note: Some HANGUP events are not generated correctly because the bridge
layer does not propagate hangupcause/hangupsource information yet.

Review: https://reviewboard.asterisk.org/r/2544/
(closes issue ASTERISK-21563)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-13 13:15:56 +00:00
Kinsey Moore
759a7e4a30 Rework stasis cache clear events
Stasis cache clear message payloads now consist of a stasis_message
representative of the message to be cleared from the cache. This allows
multiple parallel caches to coexist and be cleared properly by the same
cache clear message even when keyed on different fields.

This change fixes a bug where multiple cache clears could be posted for
channels. The cache clear is now produced in the destructor instead of
ast_hangup.

Additionally, dummy channels are no longer capable of producing channel
snapshots.

Review: https://reviewboard.asterisk.org/r/2596


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-07 12:56:56 +00:00
Richard Mudgett
6114166237 Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common transfer functions.
(closes issue ASTERISK-21523)
Reported by: Matt Jordan

(closes issue ASTERISK-21524)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2600/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-07 01:06:49 +00:00
Joshua Colp
fb6344e249 Publish the channel state snapshot *before* calling device state so a device state producer can use
an up to date snapshot.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-05 14:50:46 +00:00
Richard Mudgett
680765d452 Remove ast_channel_bridge() and associated code called only by it.
* Added some more BUGBUG notes.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-31 16:15:32 +00:00
Richard Mudgett
ccc8cc5346 Fixup hold/unhold with attended and blind transfers.
* DTMF attended and blind transfers have hold/unhold behavior restored.

* External attended and blind transfers unhold the transfered party when
the transfer is initiated.

* Made prohibit blind transferring a bridge marked as masquerade only.
(ConfBridge bridges)

* Made running an application or playing a file inside a bridge post the
hold/unhold messages if MOH is requested.

Review: https://reviewboard.asterisk.org/r/2574/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-31 15:34:20 +00:00
David M. Lee
d81c846724 Avoid unnecessary cleanups during immediate shutdown
This patch addresses issues during immediate shutdowns, where modules
are not unloaded, but Asterisk atexit handlers are run.

In the typical case, this usually isn't a big deal. But the
introduction of the Stasis message bus makes it much more likely for
asynchronous activity to be happening off in some thread during
shutdown.

During an immediate shutdown, Asterisk skips unloading modules. But
while it is processing the atexit handlers, there is a window of time
where some of the core message types have been cleaned up, but the
message bus is still running. Specifically, it's still running
module subscriptions that might be using the core message types. If a
message is received by that subscription in that window, it will
attempt to use a message type that has been cleaned up.

To solve this problem, this patch introduces ast_register_cleanup().
This function operates identically to ast_register_atexit(), except
that cleanup calls are not invoked on an immediate shutdown. All of
the core message type and topic cleanup was moved from atexit handlers
to cleanup handlers.

This ensures that core type and topic cleanup only happens if the
modules that used them are first unloaded.

This patch also changes the ast_assert() when accessing a cleaned up
or uninitialized message type to an error log message. Message type
functions are actually NULL safe across the board, so the assert was a
bit heavy handed. Especially for anyone with DO_CRASH enabled.

Review: https://reviewboard.asterisk.org/r/2562/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
Richard Mudgett
f069ee9681 Fix segfault when dealing with chan_agent channels.
Check the returned bridged pointer for NULL to avoid a crash.  It looks
like chan_agent is returning a NULL pointer when it probably should be
returning a pointer to the channel the Agent channel is pretending to be.

(closes issue ASTERISK-21793)
Reported by: Rodrigo P. Telles
Patches:
      jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Rodrigo P. Telles
........

Merged revisions 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 390047 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-29 20:24:18 +00:00
Jason Parker
fa98eb2aea Remove unused RAII vars.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-29 19:54:01 +00:00
Mark Michelson
fac3839e68 Adds support for a core attended transfer function plus adds some hiding of masquerades.
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.

The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.

Review: https://reviewboard.asterisk.org/r/2511

(closes issue ASTERISK-21334)
Reported by Matt Jordan

(closes issue Asterisk-21336)
Reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-28 14:45:31 +00:00
Jason Parker
154fbf8cae Split Hold event into Hold/Unhold, and move it into core.
(closes issue ASTERISK-21487)
Review: https://reviewboard.asterisk.org/r/2565/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 21:21:25 +00:00
Richard Mudgett
3d63833bd6 Merge in the bridge_construction branch to make the system use the Bridging API.
Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 18:00:22 +00:00
Matthew Jordan
d04f1fd60a Publish the outbound channel's application/data when dialing
This patch does two things:
* It fixes a bug where the outbound channel's application/data set by the
  dialing API/app_dial is not communicated until the channel is hung up.
  If that happens, AMI would incorrectly send a NewExten event immediately
  after a Hangup. This isn't really AMI's fault, as the dialing APIs never
  communicated the 'helpful' app/data on the outbound channel until it was
  hungup.
* It makes public sending a stasis message about a change in channel state.
  This is useful enough that - for now at least - it should be public. If
  operations on a channel go to being more coarse-grained, this function
  could be made private again.

Review: https://reviewboard.asterisk.org/r/2548

Note that this problem was found and reported by Matt DiMeo.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17 17:43:58 +00:00
David M. Lee
0eb4cf8c19 Remove required type field from channel blobs
When we first introduced the channel blob types, the JSON blobs were
self identifying by a required "type" field in the JSON object
itself. This, as it turns out, was a bad idea.

When we introduced the message router, it was useless for routing based
on the JSON type. And messages had two type fields to check: the
stasis_message_type() of the message itself, plus the type field in the
JSON blob (but only if it was a blob message).

This patch corrects that mistake by removing the required type field
from JSON blobs, and introducing first class stasis_message_type objects
for the actual message type.

Since we now will have a proliferation of message types, I introduced a
few macros to help reduce the amount of boilerplate necessary to set
them up.

Review: https://reviewboard.asterisk.org/r/2509


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 18:34:50 +00:00
Richard Mudgett
3232e23ca7 Remove the ABI compatability ast_channel_alloc(). It is no longer needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-02 19:06:53 +00:00
Richard Mudgett
c137d12111 Fix crash when AMI redirect action redirects two channels out of a bridge.
The two party bridging loops were changing the bridge peer pointers
without the channel locks held.  Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity for the
pointers to become NULL even though the masquerade code has the channels
locked.

(closes issue ASTERISK-21356)
Reported by: William luke
Patches:
      jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: William luke
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-22 16:44:21 +00:00
David M. Lee
2450722f52 DTMF events are now published on a channel's stasis_topic. AMI was
refactored to use these events rather than producing the events directly
in channel.c. Finally, the code was added to app_stasis to produce
DTMF events on the WebSocket.

The AMI events are completely backward compatible, including sending
events on transmitted DTMF, and sending DTMF start events.

The Stasis-HTTP events are somewhat simplified. Since DTMF start and
DTMF send events are generally less useful, Stasis-HTTP will only send
events on received DTMF end.

(closes issue ASTERISK-21282)
(closes issue ASTERISK-21359)
Review: https://reviewboard.asterisk.org/r/2439


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-15 16:22:03 +00:00
Matthew Jordan
b8d4e573f1 Add multi-channel Stasis messages; refactor Dial AMI events to Stasis
This patch does the following:
 * A new Stasis payload has been defined for multi-channel messages. This
   payload can store multiple ast_channel_snapshot objects along with a single
   JSON blob. The payload object itself is opaque; the snapshots are stored
   in a container keyed by roles. APIs have been provided to query for and
   retrieve the snapshots from the payload object.
 * The Dial AMI events have been refactored onto Stasis. This includes dial
   messages in app_dial, as well as the core dialing framework. The AMI events
   have been modified to send out a DialBegin/DialEnd events, as opposed to
   the subevent type that was previously used.
 * Stasis messages, types, and other objects related to channels have been
   placed in their own file, stasis_channels. Unit tests for some of these
   objects/messages have also been written.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-08 14:26:37 +00:00
Kinsey Moore
71206544a7 Break the world. Stasis message type accessors should now all be named correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-28 15:45:18 +00:00
David M. Lee
4a6237b231 Move NewCallerid, HangupRequest and SoftHangupRequest to Stasis
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.

NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.

Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.

There a a few other changes wrapped up in here as well.

 * When ast_channel_topic() is given NULL for a channel, it returns
   the ast_channel_topic_all() topic instead of NULL. This can clean
   up a lot of NULL checking we're doing currently.
 * The fields Cause and Cause-txt were removed from the base channel
   information and put only on the Hangup events, since those fields
   are meaningless outside of a Hangup event.
 * Removed the pipe-delimiter processing of the channelvars field,
   since that's been deprecated forever.

(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-25 16:19:55 +00:00
David M. Lee
cf9324b25e Move more channel events to Stasis; move res_json.c to main/json.c.
This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.

To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.

I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.

 * Move JSON support from res_json.c to main/json.c
   * Made libjansson-dev a required dependency
 * Added an ast_channel_blob message type, which has a channel
   snapshot and JSON blob of data.
 * Changed UserEvent and Newexten events so that they are dispatched
   via ast_channel_blob messages on the channel's topic.
 * Got rid of the ast_channel_varset message; used ast_channel_blob
   instead.
 * Extracted the manager functions converting Stasis channel events to
   AMI events into manager_channel.c.

(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-22 14:06:46 +00:00
Kinsey Moore
99aa02d17f Transition MWI to Stasis-core
Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.

Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16 15:45:58 +00:00
David M. Lee
c0e2ed1fe9 Ensure dummy channels get a stasis topic.
Fixes test failure introduced in r382685.

(issue ASTERISK-20887)
(issue ASTERISK-20959)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08 16:25:58 +00:00
David M. Lee
4edd8be35c This patch adds a new message bus API to Asterisk.
For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.

This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:

 - Loosely coupled; new message types can be added in seperate modules.
 - Easy to use; publishing and subscribing are straightforward
   operations.

In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.

(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-08 15:15:13 +00:00
Richard Mudgett
5b236ee647 Make ast_do_masquerade() a void function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-09 01:31:55 +00:00
Richard Mudgett
b7ecff2e4b Eliminate a use of a C++ keyword as a variable. new to new_frame
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-31 18:14:05 +00:00
Matthew Jordan
7d9871b394 Add ControlPlayback manager action
This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.

Review: https://reviewboard.asterisk.org/r/2265/

(closes issue ASTERISK-20882)
Reported by: mjordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-22 15:16:20 +00:00
David M. Lee
d3bb2506a1 Gently reduce masquerade insanity
Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.

While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.

This patch does two things to make masquerades slightly less insane:
 * When swapping the names of the original and clone channel, only emit a
   single rename event of original -> original<ZOMBIE>. The original code
   issued three rename events to accomplish the same end.
 * In addition to swapping the names of the channels, also swap their
   uniqueid's. This allows the 'Uniqueid' field to be used as a stable
   identifier for a channel from and external interface, such as AMI.

Review: https://reviewboard.asterisk.org/r/2266/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-14 15:58:01 +00:00
Richard Mudgett
2db3cc2e26 * Simplify native bridge code in ast_channel_bridge().
* Fix an unbalanced manager_bridge_event(unlink) call if
AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge().

* Make ast_channel_bridge() use common cleanup code when leaving the
bridge.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-10 02:40:50 +00:00
Richard Mudgett
e31002b9b2 * Removed some noop code and restructured an else-if ladder in ast_generic_bridge().
* Trivial changes in ast_channel_bridge().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-10 01:43:27 +00:00
Richard Mudgett
8760b2644e * Simple optimization of bridge_playfile().
* Squeezed some redundancy out of update_bridge_vars().

* Wrapped long line in __ast_change_name_nolink().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09 23:51:45 +00:00
Kinsey Moore
ad582f9aef Add missing test event
This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
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2013-01-03 18:51:43 +00:00
Matthew Jordan
8fb5bdce9a Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.

This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.

(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
  event-cachability-3.diff uploaded by jcolp (license 5000)
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2013-01-02 18:11:59 +00:00
Richard Mudgett
255d82aae3 Give the causes[] a struct name.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-20 21:51:03 +00:00
Kinsey Moore
dec9ee2f04 Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support testing
of time-limited bridges. This aids in testing the S() and L() bridge
options.

(issue SWP-4713)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-18 17:48:36 +00:00
Richard Mudgett
d7c59c19a8 Cleanup CLI commands on exit for several files.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
      unregister-cli-multiple-all.patch (license #5909) patch uploaded by Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 22:03:23 +00:00
Richard Mudgett
4ccf2c7aa5 Add red-black tree container type to astobj2.
* Add red-black tree container type.

* Add CLI command "astobj2 container dump <name>"

* Added ao2_container_dump() so the container could be dumped by other
modules for debugging purposes.

* Changed ao2_container_stats() so it can be used by other modules like
ao2_container_check() for debugging purposes.

* Updated the unit tests to check red-black tree containers.

(closes issue ASTERISK-19970)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2110/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-21 18:33:16 +00:00
Brent Eagles
6b07dd60cc Patch to prevent stopping the active generator when it is not the silence
generator.

This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.

(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-15 14:35:01 +00:00
Mark Michelson
f2bb9afe17 Multiple revisions 375993-375994
........
  r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
  
  Fix misuses of timeouts throughout the code.
  
  Prior to this change, a common method for determining if a timeout
  was reached was to call a function such as ast_waitfor_n() and inspect
  the out parameter that told how many milliseconds were left, then use
  that as the input to ast_waitfor_n() on the next go-around.
  
  The problem with this is that in some cases, submillisecond timeouts
  can occur, resulting in the out parameter not decreasing any. When this
  happens thousands of times, the result is that the timeout takes much
  longer than intended to be reached. As an example, I had a situation where
  a 3 second timeout took multiple days to finally end since most wakeups
  from ast_waitfor_n() were under a millisecond.
  
  This patch seeks to fix this pattern throughout the code. Now we log the
  time when an operation began and find the difference in wall clock time
  between now and when the event started. This means that sub-millisecond timeouts
  now cannot play havoc when trying to determine if something has timed out.
  
  Part of this fix also includes changing the function ast_waitfor() so that it
  is possible for it to return less than zero when a negative timeout is given
  to it. This makes it actually possible to detect errors in ast_waitfor() when
  there is no timeout.
  
  (closes issue ASTERISK-20414)
  reported by David M. Lee
  
  Review: https://reviewboard.asterisk.org/r/2135/
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  r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
  
  Remove some debugging that accidentally made it in the last commit.
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2012-11-07 19:15:26 +00:00
Richard Mudgett
6ad0126425 Fix stuck DTMF when bridge is broken.
When a bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending forever.

* Made simulate a DTMF end event when a bridge is broken and a DTMF digit
was in progress.

(closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy
Patches:
      bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy
      Modified to jira_asterisk_20492_v1.8.patch
      jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2169/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 19:05:11 +00:00
Matthew Jordan
a0c363e227 Refactor ast_timer_ack to return an error and handle the error in timer users
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor.  This can lead to situations where errors stream to the
CLI/log file.  This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.

This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures.  It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.

Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.

Review: https://reviewboard.asterisk.org/r/2178/

(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
  jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-05 23:10:14 +00:00
Richard Mudgett
f85db0e34d Things don't need to be that const.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-02 21:01:33 +00:00
Matthew Jordan
a094707d51 Fix a variety of ref counting issues
This patch resolves a number of ref leaks that occur primarily on Asterisk
shutdown.  It adds a variety of shutdown routines to core portions of
Asterisk such that they can reclaim resources allocate duringd initialization.

Review: https://reviewboard.asterisk.org/r/2137
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-02 01:47:16 +00:00
Mark Michelson
fdfb3ae5fa Allow for redirecting reasons to be set to arbitrary strings.
This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.

The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.

(closes issue AST-942)
reported by Malcolm Davenport

(closes issue AST-943)
reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/2101



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-25 19:29:14 +00:00
Richard Mudgett
da5944fc56 Named call pickup groups. Fixes, missing functionality, and improvements.
* ASTERISK-20383
Missing named call pickup group features:

CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.

* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up.  In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.

Regression because of the named call pickup group feature.

* See ASTERISK-20386 for the implementation improvements.  These are the
changes in channel.c and channel.h.

* Fixed some locking issues in CHANNEL().

(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2112/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20 17:22:41 +00:00
David M. Lee
192e6a0f7a Fix timeouts for ast_waitfordigit[_full].
ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds,
expecting it to decrement the timeout by however many milliseconds were
waited. This is a problem if it consistently waits less than 1ms. The timeout
will never be decremented, and we wait... FOREVER!

This patch makes ast_waitfordigit_full manage the timeout itself. It maintains
the previously undocumented behavior that negative timeouts wait forever.

(closes issue ASTERISK-20375)
Reported by: Mark Michelson
Tested by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2109/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-13 20:02:56 +00:00
Richard Mudgett
fb1d9a90a4 Enhance astobj2 to support other types of containers.
The new API allows for sorted containers, insertion options, duplicate
handling options, and traversal order options.

* Adds the ability for containers to be sorted when they are created.

* Adds container creation options to handle duplicates when they are
inserted.

* Adds container creation option to insert objects at the beginning or end
of the container traversal order.

* Adds OBJ_PARTIAL_KEY to allow searching with a partial key.  The partial
key works similarly to the OBJ_KEY flag.  (The real search speed
improvement with this flag will come when red-black trees are added.)

* Adds container traversal and iteration order options: Ascending and
Descending.

* Adds an AST_DEVMODE compile feature to check the stats and integrity of
registered containers using the CLI "astobj2 container stats <name>" and
"astobj2 container check <name>".  The channels container is normally
registered since it is one of the most important containers in the system.

* Adds ao2_iterator_restart() to allow iteration to be restarted from the
beginning.

* Changes the generic container object to have a v_method table pointer to
support other types of containers.

* Changes the container nodes holding objects to be ref counted.

The ref counted nodes and v_method table pointer changes pave the way to
allow other types of containers.

* Includes a large astobj2 unit test enhancement that tests the new
features.

(closes issue ASTERISK-19969)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/2078/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-12 21:02:29 +00:00
Jonathan Rose
595cd83d03 Masquerade: Retain parkinglot settings made by CHANNEL function.
Prior to this patch, the user would have a parkinglot set on a channel that
was parked and when the channel was retrieved, any attempt by that channel
to park would simply use the default. This patch makes parkinglot values
set in this way be retained through the masquerade.

(closes issue AST-990)
Reported by: Nick Huskinson
Patches:
    masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10 17:41:57 +00:00
Matthew Jordan
8018b879a2 Clean up doxygen warnings
This patch fixes numerous doxygen warnings across Asterisk.  It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.

Much thanks to Andrew for tackling one of the Asterisk janitor projects!

(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
  doxygen_partial.diff uploaded by Andrew Latham (license 5985)
  make_progdocs.diff uploaded by Andrew Latham (license 5985)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 14:23:28 +00:00
Richard Mudgett
7a6393d8de Fix theoretical compile error with HAVE_EPOLL.
Really shows how much epoll is used since it had not been reported yet.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 19:57:24 +00:00
Richard Mudgett
6e2d8c06ea Initialize file descriptors for dummy channels to -1.
Dummy channels usually aren't read from, but functions like SHELL and CURL
use autoservice on the channel.

(closes issue ASTERISK-20283)
Reported by: Gareth Palmer
Patches:
      svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 19:48:56 +00:00
Richard Mudgett
fb6238899b Add private representation of caller, connected and redirecting party ids.
This patch adds the feature "Private representation of caller, connected
and redirecting party ids", as previously discussed with us (DATUS) and
Digium.

1. Feature motivation

Until now it is quite difficult to modify a party number or name which can
only be seen by exactly one particular instantiated technology channel
subscriber.  One example where a modified party number or name on one
channel is spread over several channels are supplementary services like
call transfer or pickup.  To implement these features Asterisk internally
copies caller and connected ids from one channel to another.  Another
example are extension subscriptions.  The monitoring entities (watchers)
are notified of state changes and - if desired - of party numbers or names
which represent the involving call parties.  One major feature where a
private representation of party names is essentially needed, i.e.  where a
party name shall be exclusively signaled to only one particular user, is a
private user-specific name resolution for party numbers.  A lookup in a
private destination-dependent telephone book shall provide party names
which cannot be seen by any other user at any time.

2. Feature Description

This feature comes along with the implementation of additional private
party id elements for caller id, connected id and redirecting ids inside
Asterisk channels.

The private party id elements can be read or set by the user using
Asterisk dialplan functions.

When a technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting update
event, it merges the corresponding public id with the private id to create
an effective party id.  The effective party id is then used for protocol
signaling.

The channel technologies which initially support the private id
representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and
PRI (chan_dahdi).

Once a private name or number on a channel is set and (implicitly) made
valid, it is generally used for any further protocol signaling until it is
rewritten or invalidated.

To simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all connected/redirecting
update events which are generated by technology channels -- receiving
regarding protocol information - automatically trigger the invalidation of
private ids.

If not using the private party id representation feature at all, i.e.  if
using only the 'regular' caller-id, connected and redirecting related
functions, the current characteristic of Asterisk is not affected by the
new extended functionality.

3. User interface Description

To grant access to the private name and number representation from the
Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan
functions are extended by the following data types.  The formats of these
data types are equal to the corresponding regular 'non-private' already
existing data types:

CALLERID:
priv-all
priv-name priv-name-valid priv-name-charset priv-name-pres
priv-num priv-num-valid priv-num-plan priv-num-pres
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag

CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag

REDIRECTING:
priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset
priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd
priv-orig-tag

priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan
priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd
priv-from-tag

priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset
priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan
priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag

Reported by: Thomas Arimont

Review: https://reviewboard.asterisk.org/r/2030/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
Mark Michelson
eb9e645a27 Allow support for early media on AMI originates and call files.
This is based on the work done by Olle Johansson on review board.

The idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until the
channel has been answered. With this change, an EarlyMedia header can
be specified for AMI originates and an early_media option can
be specified in call files. With this option set, once early media is
received on a channel, it will be connected with the outgoing extension.

(closes issue ASTERISK-18644)
Reported by Olle Johansson

Review: https://reviewboard.asterisk.org/r/1472



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08 22:39:40 +00:00
Kinsey Moore
e571897441 Do not define a cause that doesn't actually exist
AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause
information. As such, it should not be defined and translatable as a
cause.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08 20:30:52 +00:00
Kinsey Moore
3d212da105 Add missing AST_CAUSE_* -> text translations
A few of these were missing from the list and are necessary for the Who
Hung Up? functionality.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-07 18:21:56 +00:00