Commit Graph

25480 Commits

Author SHA1 Message Date
Matthew Jordan
ef0c9fe4d8 res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP
This patch adds the following:
(1) A new module, res_hep, which implements a generic packet capture agent for
the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based
on a patch provided by Alexandr Dubovikov; I basically just wrapped it up,
added configuration via the configuration framework, and threw in a
taskprocessor.
(2) A new module, res_hep_pjsip, which forwards all SIP message traffic that
passes through the res_pjsip stack over to res_hep for encapsulation and
transmission to a HEPv3 capture server.

Much thanks to Alexandr for his Asterisk patch for this code and for a *lot*
of patience waiting for me to port it to 12/trunk. Due to some dithering on
my part, this has taken the better part of a year to port forward (I still
blame CDRs for the delay).

ASTERISK-23557 #close

Review: https://reviewboard.asterisk.org/r/3207/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-28 18:32:50 +00:00
Alexandr Anikin
a35ce3924b process stack command even if gatekeeper client isn't register
don't destroy gatekeeper client if it is not started
don't destroy gatekeeper client in some sort of gatekeeper errors
signal rtp create condition when call cleared before rtp structure created

(closes issue ASTERISK-23460)

Reported by: Dmitry Melekhov
Patches:
	ASTERISK-23460-2.patch

Tested by: Dmitry Melekhov
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2014-03-28 18:00:18 +00:00
Matthew Jordan
597f25db69 Update API versions and UPGRADE/CHANGES for 12.2.0
This patch does the following:
 * It updates the AMI version to 2.2.0 to indicate backwards compatible
   changes have been made since the last release
 * It updates the ARI version to 1.2.0 to indicate backwards compatible
   changes have been made since the last release
 * It updates the UPGRADE/CHANGES files with changes that were not
   mentioned
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2014-03-28 17:41:23 +00:00
Matthew Jordan
a438a0e65f res_config_odbc: Fix for nullable integer columns and keyfield existence check in update_odbc.
This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.

Also, the check for existence of a mandatory column checked for the first
column in the list instead of the key field lookup column. This patch fixes
that issue as well.

Finally, the compatibility option allow_empty_string_in_nontext, which was
added to previous revisions to allow for some database backends with certain
schemas to function, has been removed.

Review: https://reviewboard.asterisk.org/r/3335

ASTERISK-23459 #close
ASTERISK-23351 #close

(closes issue ASTERISK-23459)
Reported by: zvision
patches:
  res_config_odbc.diff uploaded by zvision (License 5755)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-28 17:09:14 +00:00
Matthew Jordan
24452b852d Blocked revisions 411512
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res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values.

This patch fixes setting nullable integer columns to NULL instead of an empty
string, which fails for PostgreSQL, for example. The current code is supposed
to do so, but the check is broken. The patch also allows the first column in
the list to be a nullable integer.

This patch also adds a compatibility setting in res_odbc.conf,
allow_empty_string_in_nontext. It is enabled by default. It should be disabled
for database backends (such as PostgreSQL) that require NULL instead of an
empty string for Integer columns.

Review: https://reviewboard.asterisk.org/r/3375

(issue ASTERISK-23459)
Reported by: zvision
patches:
  res_config_odbc.diff uploaded by zvision (License 5755)
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2014-03-28 16:49:09 +00:00
Scott Griepentrog
0d057e6791 http: response body often missing after specific request
This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:

a) Client request comes from node.js user agent
   "Shred" via use of swagger-client library.

b) Asterisk and Client are *not* on the same
   host or TCP/IP stack

In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function.  The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission.  See review for more details.


ASTERISK-23548 #close
(closes issue ASTERISK-23548)
Reported by: Sam Galarneau
Review: https://reviewboard.asterisk.org/r/3402/
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2014-03-28 16:18:56 +00:00
Matthew Jordan
2cfa9b4b77 UPGRADE: Note IAX2 compatibility issue between 1.4 and 1.8+ systems.
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2014-03-28 15:48:48 +00:00
Matthew Jordan
2dae3d6ea3 contrib/realtime: Remove empty SQL script files
Since the relatime scripts are now managed by Alembic, the previous realtime
scripts were previously removed. However, the removal process messed up, as
the files were still in the repository. The contents were just empty.

This removes the files from the tree.
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2014-03-28 14:19:20 +00:00
Matthew Jordan
562182f5b0 chan_sip: Add MESSAGE request to allowed methods
The allowed methods advertised by chan_sip did not previously note the MESSAGE
request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we
should advertise that we support MESSAGE requests.

ASTERISK-23504 #close
ASTERISK-23504 #comment Reported by: Martin Kontsek
ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)

Review: https://reviewboard.asterisk.org/r/3396/
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2014-03-28 03:55:26 +00:00
Corey Farrell
fbe0dfaf44 Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
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2014-03-27 19:21:44 +00:00
Corey Farrell
44409401ec main/formats: Fix crash in ast_format_cmp during non-clean shutdown.
* Update asterisk.h to reflect availability of ast_register_cleanup in 11.9.
* Use ast_register_cleanup for format_attr_shutdown.

(closes issue ASTERISK-23103)
Reported by: JoshE
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2014-03-27 18:26:12 +00:00
Mark Michelson
a8629e53c1 Give sorcery instances a reference to their wizards.
On graceful shutdown, sorcery wizards are all killed off, but it is
possible for sorcery instances to still have dangling pointers after
this, possibly causing a crash. Giving the sorcery instances a reference
to their wizards ensures that the wizard reference will remain valid for
the lifetime of the sorcery instance.

Review: https://reviewboard.asterisk.org/r/3401
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2014-03-27 14:21:15 +00:00
Joshua Colp
7dddd694cb say: Fix a bug where SayNumber in Polish tries to play incorrect sound.
This change fixes a bug where calling SayNumber with a number divisible by
100 using the Polish language would cause the code to attempt to play a
sound file with an empty name.

(closes issue ASTERISK-23509)
Reported by: zvision

Review: https://reviewboard.asterisk.org/r/3378/
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2014-03-26 22:45:10 +00:00
Jonathan Rose
fa3a2f8eca chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior too this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.

(closes issue AST-1301)
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2014-03-26 16:15:12 +00:00
Richard Mudgett
4b18b3bb4d Fix 'alembic branches' merge conflict as described by the web page.
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2014-03-26 16:05:00 +00:00
Sean Bright
c32fe8b8e5 ARI: Don't complain about missing ARI users when we aren't enabled
Currently, if ARI is not enabled it will still complain that there are no
configured users.  This patch checks to see if ARI is enabled before logging and
error or iterating the container to validate the users.

Review: https://reviewboard.asterisk.org/r/3391/
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2014-03-25 18:44:57 +00:00
Mark Michelson
2bf37a417d Add a "message_context" option for PJSIP endpoints.
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2014-03-25 17:40:51 +00:00
Richard Mudgett
c1c8300e27 res_pjsip: Fix contact authenticate_qualify endpoint lookup when qualifing a contact.
* Fixed bad use of ao2_find() in on_endpoint().

* Replaced use of find_endpoints() with find_an_endpoint() since only the
first found endpoint is ever needed.

* Fixed qualify_contact_cb() to update the contact with the aor
authenticate_qualify setting.  Otherwise, permanent contacts in the aor
type sections would have a config line order dependancy.

* Fixed off nominal path contact ref leak in qualify_contact().  The
comment saying the unref is not needed was wrong.

* Fixed off nominal path use of the endpoint parameter if it is NULL in
send_out_of_dialog_request().

* Added missing off nominal path unref of pjsip tdata in
send_out_of_dialog_request().

* Fixed off nominal path failing to call the callback in send_request_cb()
when the request is challenged for authentication.

* Eliminated silly RAII_VAR() use in qualify_contact_cb().

* Updated ast_sip_send_request() doxygen to better reflect reality.

(closes issue ASTERISK-23254)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/3381/
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2014-03-25 16:57:41 +00:00
Kinsey Moore
a4890eddfb chan_sip: Fix incorrect use of timers
If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.

(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
    provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
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2014-03-25 16:06:57 +00:00
Jonathan Rose
eb0a982f8c ARI: Resolve a subscription leak against implicit bridge subscriptions
When a channel in a stasis application is joined to a bridge, a subscription
for that bridge is created implicitly for the stasis application serving the
channel. Prior to this patch, subsequent removals of the channel from the
bridge would leave the subscription open.

Review: https://reviewboard.asterisk.org/r/3380/
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2014-03-25 15:56:05 +00:00
Richard Mudgett
c8ebf3e3c7 Revert -r411073. It didn't help and blew up the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-25 15:47:17 +00:00
Richard Mudgett
89e12de79d locking: Add temporary sanity checks.
Add some temporary sanity checks to hunt for locking problems with the
masquerade supertest.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-24 23:36:36 +00:00
Joshua Colp
6d81951f0d chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted.
(closes issue ASTERISK-20841)
Reported by: Kelly Goedert
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2014-03-24 21:39:46 +00:00
Richard Mudgett
236d17362d res_pjsip_registrar.c: Miscellaneous cleanup in rx_task().
* Fix variable shadowing of 'updated' by renaming it to 'contact_update'.

* Checked 'contact_update' for ast_sorcery_copy() failure.

* Removed silly use of RAII_VAR() for 'contact_update'.
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2014-03-21 16:04:09 +00:00
Sean Bright
b44d324891 Make the AEL load process less chatty.
Switched a bunch of LOG_NOTICEs to ast_debug.  This time without breaking the
build.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:50:11 +00:00
Sean Bright
14942ecb17 Revert r410981. aelparse blew up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:30:37 +00:00
Sean Bright
df2d959d7d Remove a LOG_NOTICE from ast_config_engine_register.
There is enough indication from the CLI that we are loading a realtime engine
as it is.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:16:50 +00:00
Sean Bright
922d0b7565 Make the AEL load process less chatty.
Switched a bunch of LOG_NOTICEs to ast_debug.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:14:13 +00:00
Jonathan Rose
3c16865fc2 app_confbridge: Fix bug - users with startmuted set don't start muted
(closes issue ASTERISK-23461)
Reported by: Chico Manobela
Review: https://reviewboard.asterisk.org/r/3373/
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2014-03-20 23:02:45 +00:00
Richard Mudgett
1ba13718fc assigned-uniqueids: Miscellaneous cleanup and fixes.
* Fix memory leak in ast_unreal_new_channels().  Made it generate the ;2
uniqueid on a stack variable instead of mallocing it.

* Made send error response to ARI and AMI requests instead of just logging
excessive uniqueid length and allowing truncation.  action_originate() and
ari_channels_handle_originate_with_id().

* Fixed minor truncating uniqueid hole when generating the ;2 uniqueid
string length.  Created public and internal lengths of uniqueid.  The
internal length can handle a max public uniqueid plus an appended ;2.

* free() and ast_free() are NULL tolerant so they don't need a NULL test
before calling.

* Made use better struct initialization format instead of the position
dependent initialization format.  Also anything not explicitly initialized
in the struct is initialized to zero by the compiler.

* Made ast_channel_internal_set_fake_ids() use the safer
ast_copy_string() instead of strncpy().

Review: https://reviewboard.asterisk.org/r/3371/
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2014-03-20 16:35:57 +00:00
Mark Michelson
57239bfe37 PJSIP: Allow for identify sections to be specified in sorcery.conf.
"identify" is a special type of configuration object in PJSIP because
unlike the other objects, it is not provided by the base res_pjsip module.
Instead, it is provided by the res_pjsip_endpoint_identifier_ip module. If
using the default sorcery wizard (config,criteria=type=identify) then things
work because the module that applies the default wizard is the correct module.

However, if attempting to use sorcery.conf to apply an alternate wizard, it
was not possible. If you attempted to specify the identify object type in the
res_pjsip section, then the object could not be registered since the object
was undocumented for the res_pjsip module. There was no alternate configuration
section defined for it, so you were out of luck if you wanted to override the
default wizard.

With this change, the identify section will properly have a sorcery.conf-based
wizard applied when the identify definition is within the res_pjsip_endpoint_identifier_ip
section.
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2014-03-19 17:27:57 +00:00
Joshua Colp
326153d949 res_stasis: Fix a bug where the default bridge type was not set.
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2014-03-19 14:25:31 +00:00
Joshua Colp
1cf74b8776 res_stasis: Extend bridge type to be a comma separated list of bridge attributes.
This change turns the bridge type field into a comma separated list of attributes.
These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting
the various attributes a user can control the type of bridge created with the
behavior they need for their application.

(closes issue ASTERISK-23437)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3359/
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2014-03-19 12:54:25 +00:00
Matthew Jordan
e33e003f78 res_ari: Fix documentation schema error
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2014-03-19 02:33:55 +00:00
Rusty Newton
35fb3a564b res_ari: Add notes about Asterisk HTTP server to the "enabled" config option for the res_ari general section
Added note and see-also reminding user to enable the HTTP server.

(closes issue ASTERISK-22499)
Reported by: Rusty Newton
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Merged revisions 410876 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-18 23:32:00 +00:00
Scott Griepentrog
eecb74a9a7 ARI: allow json content type with zero length body
When a request was received with a Content-type of json,
the body was sent for json parsing - even if it was zero
length.  This resulted in ARI requests failing that were
valid, such as a channel DELETE with no parameters.  The
code has now been changed to skip json parsing with zero
content length.

(closes issue SWP-6748)
Reported by: Samuel Galarneau
Review: https://reviewboard.asterisk.org/r/3360/
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Merged revisions 410858 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-18 15:45:04 +00:00
Matthew Jordan
77db94a25a cdr: Add asserts for when we don't know about a CDR for a channel
In the CDR core, every channel should either be filtered out (due to being an
'internal' channel used as an implementation detail, such as playing media
back into a bridge) or it should get a CDR. Even if that CDR ends up being
discarded, we still give the channel a CDR in case we end up needing it. If we
hit a situation where a channel does not have a CDR, we should blow up in
-dev-mode. Asserts are appropriate for that.

This patch adds those asserts, as they would have quickly caught the error
fixed by r410814.
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Merged revisions 410861 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-18 15:28:45 +00:00
Joshua Colp
216b04e6f4 res_pjsip: Fix memory leak of nameservers in off-nominal resolver creation failure.
Thanks Walter Doekes!
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Merged revisions 410844 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-18 12:45:49 +00:00
Sean Bright
8357027080 res_fax_spandsp: Use g711_free() when available.
Per Johann Steinwendtner on the asterisk-dev mailing list:

http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html

g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a
noop.  I opted not to remove the call to g711_release() since it is harmless
and to call g711_free() if we have a sufficiently recent version of spandsp.

(issue ASTERISK-20149)
Reported by: Alexandr Gordeev
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Merged revisions 410829 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410830 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-18 11:52:15 +00:00
Richard Mudgett
614b6abc38 stasis_cache: Use the right variable in the cache entry ao2 cmp function.
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Merged revisions 410813 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-18 02:09:25 +00:00
Joshua Colp
cc40bf5317 res_pjsip: Enable PJSIP DNS client support.
This change enables DNS client support within PJSIP. System
nameservers are automatically discovered using res_init or
res_ninit. If this fails then PJSIP will resort to using
gethostbyname for resolution.

By enabling this support we gain SRV support, failover, and
weight support.

(closes issue ASTERISK-23435)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3343/
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Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-17 22:54:32 +00:00
Joshua Colp
932fb5a6e2 res_pjsip_multihomed: Make address replacement less aggressive.
This change makes the res_pjsip_multihomed module less aggressive when
changing the address in messages. It will now only occur if the transport
in use is bound to the any address OR if the system determined source
address matches the bound address of the transport in use.

Review: https://reviewboard.asterisk.org/r/3369/
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Merged revisions 410793 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-17 22:46:56 +00:00
Russ Meyerriecks
ed50ef4dc8 callerid: Logic error in checksum processing
Callerid checksum-ing was being handled incorrectly here. When the checksum is
calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This
value will then fail the otherwise correct callerid message.

This patch changes the logic to simply add the calculated checksum to the
transmitted 2's compliment checksum.  

Review: https://reviewboard.asterisk.org/r/3356/
(closes issue ASTERISK-23488)

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This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12
I didn't want a broken patch to be comitted to trunk so I pre-merge merged them.


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2014-03-17 22:24:03 +00:00
Mark Michelson
eba91d2a98 Revert changes to sorcery that accidentally got committed.
These changes were still up for review and have not been approved
yet. I must have had the changes in my working copy when making
a different change.
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Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-17 19:35:17 +00:00
Mark Michelson
d44aefeef4 Fix stuck channel in ARI through the introduction of synchronous bridge actions.
Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.

The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
  would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
  playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
  not to occur (the bridge dies, the channel is removed from the bridge), then we would
  never be notified.

The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.

Review: https://reviewboard.asterisk.org/r/3338
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Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-17 17:22:12 +00:00
Richard Mudgett
1900bae7b6 app_confbridge: Add missing destructor call to announcer channel destructor.
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Merged revisions 410671 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-17 16:48:55 +00:00
Matthew Jordan
2c5484c869 stasis/app.c: Add some extra debugging for subscription counts
Events are sent to a connected ARI application based on the things that ARI
application cares about. These subscriptions can be set up implicitly - such
as when that ARI application creates a new object - or explicitly, via the
application resource's subscription operations. Debugging *why* something was
being sent to an application - or why something was not being sent to an
application - was a bit tricky, as there was no debug information for the
subscriptions.

This patch adds some debug level 3 statements that show the subscription counts
for applications. (Level 3 was chosen as it matches the verbose level 3
statements elsewhere)
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Merged revisions 410650 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-16 20:27:28 +00:00
Russell Bryant
a0a51a65e0 framehook.h: Fix some doc typos.
There were a number of instances in this header file where "function all" was
intended to be "function call".  This patch fixes that up.


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2014-03-15 15:24:23 +00:00
Mark Michelson
478c7faf0b Fix failing realtime sorcery tests.
The store realtime callback needs to return a positive value for
sorcery to treat the store as a success.
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Merged revisions 410625 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-14 21:56:21 +00:00
Jonathan Rose
3a565767d7 manager: fix memory leak in manager_add_filter function
(closes issue ASTERISK-23420)
Reported by: Etienne Lessard
Patches:
    manager_eventfilter_leak uploaded by Etienne Lessard (license 6394)
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Merged revisions 410609 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410623 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-14 21:36:55 +00:00