Commit Graph

73 Commits

Author SHA1 Message Date
Nathan Bruning
10292dc663 app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions
Add a new "masquarade" channel event, and use it in app_queue to track unique id's.

Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210

ASTERISK-28829 #close
ASTERISK-25844 #close

Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6
2020-05-05 10:09:27 -05:00
Kevin Harwell
f145b58542 various modules: json integer overflow
There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:

unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);

would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.

ASTERISK-28480

Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1
2019-08-01 16:22:01 -05:00
sungtae kim
707897e5f7 main/json.c: Added app_name, app_data to channel type
It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.

ASTERISK-28343

Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c
2019-04-05 02:28:25 +02:00
Joshua Colp
07b3253155 stasis: Improve topic/subscription names and statistics.
Topic names now follow: <subsystem>:<functionality>[/<object>]

This ensures that they are all unique, and also provides better
insight in to what each topic is for.

Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.

Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.

Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.

ASTERISK-28335

Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d
2019-03-11 11:39:08 -03:00
Corey Farrell
0a9904e1c6 astobj2: Eliminate usage of legacy ao2_container_alloc routine.
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.

ao2_container_alloc is now restricted to modules only and is being
removed from Asterisk 17.

Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
2018-11-21 09:56:07 -05:00
Richard Mudgett
751fd9b628 stasis_channels.c: Misc cleanup.
* Use current OBJ_SEARCH_xxx defines instead of the deprecated versions.

* Fix hash_cb and cmp_cb container functions to correctly use the
OBJ_SEARCH_xxx values.

* Remove incorrect usage of CMP_STOP.  Most uses in the system have no
effect.  This allows the collapse of channel_role_single_cmp_cb() and
channel_role_multi_cmp_cb() into channel_role_cmp_cb().

* Remove unnecessary usage of RAII_VAR().

Change-Id: I02c405518cab22aa2a082b61e2353bf7cd629a70
2017-12-28 13:37:31 -06:00
Richard Mudgett
28eddc7ea6 stasis_channels.c: Don't set channel snapshot caller_dnid twice.
Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471
2017-12-11 14:14:16 -06:00
Richard Mudgett
6bc8af2433 stasis_channels.c: Remove a very silly RAII_VAR().
Change-Id: I28b458b3c1a442c4ef0be7b4986a95ea4149e14f
2017-11-06 11:58:57 -06:00
Joshua Colp
a9d6fc571d json: Audit ast_json_* usage for thread safety.
The JSON library Asterisk uses, jansson, is not thread
safe for us in a few ways. To help with this wrappers for JSON
object reference count increasing and decreasing were added
which use a global lock to ensure they don't clobber over
each other. This does not extend to reference count manipulation
within the jansson library itself. This means you can't safely
use the object borrowing specifier (O) in ast_json_pack and
you can't share JSON instances between objects.

This change removes uses of the O specifier and replaces them
with the o specifier and an explicit ast_json_ref. Some cases
of instance sharing have also been removed.

ASTERISK-25601 #close

Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1
2015-12-16 17:17:30 -04:00
Kevin Harwell
f5d5aa67dc AMI: Escape string values.
So this issue is a bit complicated. Since it is possible to pass values to AMI
that contain a '\r\n' (or other similar sequences) these values need to be
escaped. One way to solve this is to escape the values and then pass the escaped
values to the AMI variable parameter string building function. However, this
puts the onus on the pre-build function to escape all string values. This
potentially requires a fair amount of changes along with a lot of string
allocations/freeing for all values.

Surely there is a way to push this complexity down a level into the string
building function itself? This of course is possible, but ends up requiring a
way to distinguish between strings that need to be escaped and those that don't.
The best way to handle this is by introducing a new format specifier in the
format string. For instance a %s (no escape) and %S (escape). However, that is
a bit weird and unexpected.

So faced with those possibilities this patch implements a limited version of the
first option. Instead of attempting to escape all string values this patch only
escapes those values that make sense. This approach limits the number of changes
and doesn't suffer from the odd format specifier problem.

ASTERISK-24934 #close
Reported by: warren smith

Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0
2015-06-03 14:03:18 -05:00
Joshua Colp
181ae3b8d9 stasis: Fix dial masquerade datastore lifetime
A recent change went into Asterisk which added reference counts to the
channels stored in a dial masquerade datastore. Unfortunately this
included a reference to the caller in a dialing operation. While all
of the dialed targets have the datastore removed from them upon dialing
completion this did not occur for the caller, causing it to have a
reference to itself that could go never go away (as it depended on
the destruction of the datastore which only happened when the channel
was destroyed). This resulted in the caller channel remaining on the
system despite it having hung up.

This change does the following to fix this issue:

1. The dial masquerade datastore is now removed from the caller upon
dialing completion, just like the dialed targets.
2. Upon destruction of the caller all the dialed targets are also
removed from the dial masquerade datastore (just in case).
3. The reference to the caller has been removed as it should not be
possible for the datastore to now be valid/useful after the lifetime
of the caller has ended.

ASTERISK-25025 #close

Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f
2015-05-05 07:34:19 -03:00
Mark Michelson
077979618b Prevent potential crash on blond transfer.
Scenario:
Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects
the incoming call (or some other immediate circumstance causes Carol not
to answer the call)

What occurs in this case is that when the bridge between Alice and Bob
breaks, Alice is told to masquerade into Bob's channel that had placed
the call to Carol. The actual masquerade goes down without a hitch.
However, a channel fixup callback that attempts to publish dial events
over Stasis has a crash. The reason for this crash is that the datastore
on Bob's channel that placed the outbound call to Carol only had a bare
pointer to Carol's channel. Since Carol rejected the incoming call,
Carol's channel has been hung up and freed, meaning accessing her
channel results in a crash.

The fix here is simple. The dial fixup code has been altered to hold
references to the involved channels and to drop those references when
freeing data.

ASTERISK-25025 #close
Reported by Chet Stevens

Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197
2015-04-30 15:42:01 -05:00
Matthew Jordan
ab803ec342 ARI: Add the ability to intercept hold and raise an event
For some applications - such as SLA - a phone pressing hold should not behave
in the fashion that the Asterisk core would like it to. Instead, the hold
action has some application specific behaviour associated with it - such as
disconnecting the channel that initiated the hold; only playing MoH to channels
in the bridge if the channels are of a particular type, etc.

One way of accomplishing this is to use a framehook to intercept the
hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
accomplishes that using a new dialplan function, HOLD_INTERCEPT.

In addition, some general cleanup of raising hold/unhold Stasis messages was
done, including removing some RAII_VAR usage.

Review: https://reviewboard.asterisk.org/r/4549/

ASTERISK-24922 #close


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-07 15:21:17 +00:00
Joshua Colp
bc357c1d7e core: Don't create snapshots with locks.
Snapshots are immutable and are never changed. Allocating them
with a lock is wasteful.

Review: https://reviewboard.asterisk.org/r/4469/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-10 23:09:17 +00:00
Mark Michelson
43dd42d8ae Fix some memory leaks.
These memory leaks were found and fixed by John Hardin. I'm just
committing them for him.

ASTERISK-24736 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/4389



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-30 16:47:50 +00:00
Kevin Harwell
2f21f85c37 ARI/AMI: Include language in standard channel snapshot output
The channel "language" was already part of a channel snapshot, however is was
not sent out over AMI or ARI. This patch makes it so the channel "language" is
included in the appropriate AMI or ARI events.

ASTERISK-24553 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4245/
........

Merged revisions 429204 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-12-09 20:19:40 +00:00
Mark Michelson
93b1df3bf6 Add new AMI and ARI events for connected line changes on a channel.
The AMI event is called NewConnectedLine and the ARI event is called
ChannelConnectedLine.

ASTERISK-24554 #close
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/4231



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-08 15:49:24 +00:00
Richard Mudgett
9a7fb8ee6c stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2)
Masquerades into and out of channels that are involved in a dial operation
don't create the expected dial end event.  The missing dial end event goes
against the model for things like CDRs and generating Dial end manager
actions and such.

There are four cases:

1) A channel masquerades into the caller channel.  The case happens when
performing a blonde transfer using the channel driver's protocol.

2) A channel masquerades into a callee channel.  The case happens when
performing a directed call pickup.

3) The caller channel masquerades out of dial.  The case happens when
using the Bridge application on the caller channel.

4) A callee channel masquerades out of dial.  The case happens when using
the Bridge application on a peer channel.

As it turned out, all four cases need to be handled instead of just the
first one.

ASTERISK-24237
Reported by: Richard Mudgett

ASTERISK-24394 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4066/
........

Merged revisions 425430 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-10-14 16:35:11 +00:00
Jonathan Rose
d1b1e911bf Stasis_channels: Resolve unfinished Dials when doing masquerades
Masquerades into channels that are in the dialing state don't end their dial
and this goes against the model for things like CDRs and generating Dial end
manager actions and such.

ASTERISK-24237 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3990/
........

Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-09-19 15:18:01 +00:00
Mark Michelson
ef70c08dc7 Improve call forwarding reporting, especially with regards to ARI.
This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11 18:32:37 +00:00
Kinsey Moore
f1036f40dc Stasis: Allow message types to be blocked
This introduces stasis.conf and a mechanism to prevent certain message
types from being published. Internally, this works by preventing the
chosen message types from being created which ensures that those
message types can never be published. This patch also adjusts message
publishers such that message payloads are not created if the related
message type is not available.

ASTERISK-23943 #close
Review: https://reviewboard.asterisk.org/r/3823/


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2014-08-06 12:55:28 +00:00
Matthew Jordan
a965613cf7 stasis_channels: Update the stasis cache if manager variables are needed
In r416211, the publishing of variable changes was modified such that a
cached channel snapshot was used if manager variables were not requested
with each AMI event. This was done to reduce the amount of channel snapshots
created.

However, an assumption was made that generating a channel snapshot and
publishing the snapshot to the channel topic was sufficient to ensure that
the cache would be updated; this is not the case. The channel snapshot type
must be used to force a snapshot update.

This patch updates the publication of channel variables such that the cache
is updated prior to publication of the channel variable message if manager
variables are in use. This ensures that all AMI events receive the variable
update when they are supposed to.

Note that this issue was caught by the Asterisk Test Suite (go go testing)
........

Merged revisions 416557 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-06-18 04:22:05 +00:00
Matthew Jordan
9cc1a8e893 stasis: Reduce creation of channel snapshots to improve performance
During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
 * AGI execution
 * Returning objects for ARI commands
 * During some Local channel operations
 * During some dialling operations
 * During variable setting
 * During some bridging operations
And more.

This patch does the following:
 - It removes a number of fields from channel snapshots. These fields were
   rarely used, were expensive to have on the snapshot, and hurt performance.
   This included formats, translation paths, Log Call ID, callgroup, pickup
   group, and all channel variables. As a result, AMI Status,
   "core show channel", "core show channelvar", and "pjsip show channel" were
   modified to either hit the live channel or not show certain pieces of data.
   While this is unfortunate, the performance gain from this patch is worth
   the loss in behaviour.
 - It adds a mechanism to publish a cached snapshot + blob. A large number of
   publications were changed to use this, including:
   - During Dial begin
   - During Variable assignment (if no AMI variables are emitted - if AMI
     variables are set, we have to make snapshots when a variable is changed)
   - During channel pickup
   - When a channel is put on hold/unhold
   - When a DTMF digit is begun/ended
   - When creating a bridge snapshot
   - When an AOC event is raised
   - During Local channel optimization/Local bridging
   - When endpoint snapshots are generated
   - All AGI events
   - All ARI responses that return a channel
   - Events in the AgentPool, MeetMe, and some in Queue
 - Additionally, some extraneous channel snapshots were being made that were
   unnecessary. These were removed.
 - The result of ast_hashtab_hash_string is now cached in stasis_cache. This
   reduces a large number of calls to ast_hashtab_hash_string, which reduced
   the amount of time spent in this function in gprof by around 50%.

#ASTERISK-23811 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3568/
........

Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
Matthew Jordan
53968c00b3 TALK_DETECT: A channel function that raises events when talking is detected
This patch adds a new channel function TALK_DETECT that, when set on a
channel, causes events indicating the start/stop of talking on a channel to be
emitted to both AMI and ARI clients. 

The function allows setting both the silence threshold (the length of silence
after which we decide no one is talking) as well as the talking threshold (the
amount of energy that counts as talking). Parameters can be updated on a channel
after talk detection has been enabled, and talk detection can be removed at
any time.

The events raised by the function use a nomenclature similar to existing AMI/ARI
events.
For AMI: ChannelTalkingStart/ChannelTalkingStop
For ARI: ChannelTalkingStarted/ChannelTalkingFinished

Review: https://reviewboard.asterisk.org/r/3563/

#ASTERISK-23786 #close
Reported by: Matt Jordan
........

Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-05-30 12:42:57 +00:00
Scott Griepentrog
cf21644d6a ARI: Add ability to raise arbitrary User Events
User events can now be generated from ARI.  Events can be signalled with
arbitrary json variables, and include one or more of channel, bridge, or
endpoint snapshots.  An application must be specified which will receive
the event message (other applications can subscribe to it).  The message
will also be delivered via AMI provided a channel is attached.  Dialplan
generated user event messages are still transmitted via the channel, and
will only be received by a stasis application they are attached to or if
the channel is subscribed to.

This change also introduces the multi object blob mechanism used to send
multiple snapshot types in a single message.  The dialplan app UserEvent
was also changed to use multi object blob, and a new stasis message type
created to handle them.

ASTERISK-22697 #close
Review: https://reviewboard.asterisk.org/r/3494/
........

Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-05-22 16:09:51 +00:00
Richard Mudgett
cbe7f65674 app_dial and app_queue: Make lock the forwarding channel while taking the channel snapshot.
* Fixed ast_channel_publish_dial_forward() not locking the forwarded
channel when taking the channel snapshot.

* Fixed app_dial.c:do_forward() using the wrong channel to get the
original call forwarding string.

* Removed unnecessary locking when calling ast_channel_publish_dial() and
ast_channel_publish_dial_forward() in app_dial and app_queue.  Holding
channel locks when calling ast_channel_publish_dial_forward() with a
forwarded channel could result in pausing the system while the stasis bus
completes processsing a forwarded channel subscription.

Review: https://reviewboard.asterisk.org/r/3451/
........

Merged revisions 412579 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-04-18 16:27:31 +00:00
Richard Mudgett
ba1db5d8f5 Eliminate some more unnecessary RAII_VAR() uses.
RAII_VAR() is not a hammer appropriate to pound all nails.
........

Merged revisions 412413 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-15 18:30:24 +00:00
Richard Mudgett
45ade68cb4 Remove unused RAII_VAR() declarations.
* Remove unused RAII_VAR() declarations.  The compiler cannot catch these
because the cleanup function "references" the unused variable.  Some
actually allocated and released resources that were never used.

* Fixed some whitespace issues in stasis_bridges.c.
........

Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-04-15 18:01:47 +00:00
Richard Mudgett
7542361f4a stasis_channels.c: Eliminate another overuse of RAII_VAR().
........

Merged revisions 411636 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-04-01 22:34:30 +00:00
Richard Mudgett
d277f3ec3e json: Fix off-nominal json ref counting issues.
* Fixed off-nominal json ref counting issue with using the following API
calls: ast_json_object_set() and ast_json_array_append().

* Fixed off-nominal error reporting in ast_ari_endpoints_list().

* Fixed some miscellaneous off-nominal json ref counting issues in
report_receive_fax_status() and dial_to_json().
........

Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-02-21 18:04:54 +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
........

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
3a5cc054ed res_stasis: Expose event for call forwarding and follow forwarded channel.
This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.

(closes issue ASTERISK-22719)
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/3054/
........

Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-14 17:19:41 +00:00
David M. Lee
1212906351 Reverting r403311. It's causing ARI tests to hang.
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Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03 17:07:29 +00:00
Kinsey Moore
d9015a5356 ARI: Don't leak implementation details
This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.

This prevents unhelpful error messages from being generated by
ast_json_pack.

This also corrects a bug where BridgeCreated events would not be
created.

(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
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Merged revisions 403069 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-22 20:10:46 +00:00
Matthew Jordan
aff0faf6ba stasis_channels: Don't give preference to ANI info in channel snapshots
When publishing channel snapshots, we currently compute the caller ID name and
number by giving preference first to ani.{name|number}, then to
id.{name|number}. However, when a channel driver (such as chan_sip) updates the
caller ID, it typically only updates the caller ID stored in id.{name|number}.
This means that we are currently giving preference to stale information.

When looking at the rest of the code base, the only other place where we appear
to use this same logic is in app_amd. Everywhere else, we treat the party
information in ani as being separate to the party information in id.

This patch publishes only the caller ID name and number in the snapshot field
for caller_name and caller_num. Note that the information in ANI is still
available in caller_ani.

Review: https://reviewboard.asterisk.org/r/2992/
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Merged revisions 402501 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-05 20:59:39 +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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Merged revisions 400265 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02 16:23:34 +00:00
Matthew Jordan
3d9c5e61af Fix error in using ast_channel_snapshot_type before initialization
Starting Asterisk would kick back an ERROR message stating that the Stasis
message type ast_channel_snapshot_type was used prior to initialization.
This occurred due to the caching topic being created prior to the message
type that it depended on.

This patch re-orders the start up such that the message type is initialized
prior to the caching topic. It also checks the return value of the
initialization of the agent login/logoff types.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23 20:14:46 +00:00
Richard Mudgett
e47d3db365 Doxygen comment tweaks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16 17:33:21 +00:00
David M. Lee
f29d969a79 Stasis: address refcount races; implementation comments
Change r395954 reordered some stasis object destruction, which should
have been fine. Unfortunately, it caused some hard to reproduce issues
related to objects being accessed after they had been destroyed. The
patch in r396329 fixed the destruction order problem; this patch
addresses the underlying issue. A few other stasis-related fixes were
also added.

 * Add ref-bumps around areas where objects may get transitively
   destroyed. (For example, where we lock a topic, unref a subscription,
   which unrefs the topic, which explodes the topic when we try to
   unlock it.)

 * Wrote an extensive doxygen page about Stasis implementation,
   relationships between objects, lifecycles of objects, how the
   refcounting works, etc. Many other comments were added, corrected, or
   cleaned up.

 * Added an assert to the topic dtor to catch extra ref decrements.

 * Fixed type used after destruction errors for graceful shutdown in
   stasis_channels.c.

 * I added two unit tests in an attempt to catch destruction order
   issues. Since the underlying cause is a race condition, though, the
   tests rarely failed even when the code was wrong.

 * Fixed a leak in stasis_cache_pattern.c.

(closes issue ASTERISK-22243)
Review: https://reviewboard.asterisk.org/r/2746/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16 16:03:34 +00:00
Kinsey Moore
03090a88ba Fix documentation replication issues
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
2013-08-01 17:07:52 +00:00
David M. Lee
e1b959ccbb Split caching out from the stasis_caching_topic.
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
2013-08-01 13:49:34 +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
Kinsey Moore
5a8f32703c Filter channels used as internal mechanisms
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
2013-07-19 19:23:39 +00:00
Richard Mudgett
d43b17a872 Replace chan_agent with app_agent_pool.
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
2013-07-15 23:20:55 +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
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
Matthew Jordan
4123b27f6f Remove parkinglot from the channel snapshot
Legacy channel drivers often include the ability to set a default parking lot
on an endpoint basis; when channels are created for that endpoint, they inherit
the parkinglot option. Parking used to use this option more frequently; while
it is still supported, other options (such as using channel variables or
creation of a custom parkinglot) are supported. More importantly, conveying the
parkinglot information through a channel snapshot isn't terribly useful - it
is rarely (if ever) changed on a channel and some consumers of channel
snapshots, such as ARI, will never use the information.

(closes issue ASTERISK-21968)
Reported by: Matt Jordan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 03:08:58 +00:00
Richard Mudgett
02f55a36a0 Revert accidental overcommit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03 23:57:37 +00:00
Richard Mudgett
b4e9a3fc2f Add BUGBUG note for ASTERISK-22009
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03 23:55:53 +00:00