[asterisk-commits] file: branch file/bridge_unreal_optimizer r411869 - in /team/file/bridge_unre...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Apr 7 11:05:49 CDT 2014
Author: file
Date: Mon Apr 7 11:05:17 2014
New Revision: 411869
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411869
Log:
Multiple revisions 410830,410844,410858,410861,410876,410890,410904,410918,410933,410949,410966,410995,411023,411086,411091,411141,411157,411159,411173,411191,411193,411245,411295,411311,411315,411374,411426,411442,411459,411461,411465,411512,411514,411529,411532,411534,411586,411634,411636,411638,411668,411670,411687,411701,411717,411790,411804,411809,411811
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r410830 | seanbright | 2014-03-18 08:51:24 -0300 (Tue, 18 Mar 2014) | 16 lines
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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r410844 | file | 2014-03-18 09:45:01 -0300 (Tue, 18 Mar 2014) | 4 lines
res_pjsip: Fix memory leak of nameservers in off-nominal resolver creation failure.
Thanks Walter Doekes!
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r410858 | sgriepentrog | 2014-03-18 11:51:02 -0300 (Tue, 18 Mar 2014) | 14 lines
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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r410861 | mjordan | 2014-03-18 12:28:13 -0300 (Tue, 18 Mar 2014) | 12 lines
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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r410876 | newtonr | 2014-03-18 20:31:05 -0300 (Tue, 18 Mar 2014) | 7 lines
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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r410890 | mjordan | 2014-03-18 23:29:27 -0300 (Tue, 18 Mar 2014) | 2 lines
res_ari: Fix documentation schema error
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r410904 | file | 2014-03-19 09:52:55 -0300 (Wed, 19 Mar 2014) | 12 lines
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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r410918 | file | 2014-03-19 11:24:43 -0300 (Wed, 19 Mar 2014) | 2 lines
res_stasis: Fix a bug where the default bridge type was not set.
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r410933 | mmichelson | 2014-03-19 14:26:22 -0300 (Wed, 19 Mar 2014) | 20 lines
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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r410949 | rmudgett | 2014-03-20 13:27:49 -0300 (Thu, 20 Mar 2014) | 25 lines
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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r410966 | jrose | 2014-03-20 19:54:59 -0300 (Thu, 20 Mar 2014) | 9 lines
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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r410995 | rmudgett | 2014-03-21 13:01:56 -0300 (Fri, 21 Mar 2014) | 8 lines
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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r411023 | file | 2014-03-24 18:38:21 -0300 (Mon, 24 Mar 2014) | 11 lines
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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r411086 | jrose | 2014-03-25 12:44:29 -0300 (Tue, 25 Mar 2014) | 9 lines
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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r411091 | kmoore | 2014-03-25 13:04:10 -0300 (Tue, 25 Mar 2014) | 21 lines
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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r411141 | rmudgett | 2014-03-25 13:55:16 -0300 (Tue, 25 Mar 2014) | 32 lines
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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r411157 | mmichelson | 2014-03-25 14:30:56 -0300 (Tue, 25 Mar 2014) | 3 lines
Add a "message_context" option for PJSIP endpoints.
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r411159 | mmichelson | 2014-03-25 14:52:39 -0300 (Tue, 25 Mar 2014) | 15 lines
Prevent duplicate sorcery wizards from being applied to sorcery object types.
This commit contains several changes to sorcery:
1) Application of sorcery configuration based on module name is automatically performed
when sorcery is opened for a module.
2) Sorcery will not attempt to apply the same wizard to an object type more than once.
3) Sorcery gives more exact results when attempting to apply a wizard, whether as the
default or based on configuration.
Sorcery unit tests still pass for me after making these changes.
Review: https://reviewboard.asterisk.org/r/3326
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r411173 | seanbright | 2014-03-25 15:43:57 -0300 (Tue, 25 Mar 2014) | 8 lines
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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r411191 | rmudgett | 2014-03-26 13:03:24 -0300 (Wed, 26 Mar 2014) | 1 line
Fix 'alembic branches' merge conflict as described by the web page.
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r411193 | jrose | 2014-03-26 13:07:31 -0300 (Wed, 26 Mar 2014) | 17 lines
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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r411245 | file | 2014-03-26 19:44:40 -0300 (Wed, 26 Mar 2014) | 17 lines
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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r411295 | mmichelson | 2014-03-27 11:20:10 -0300 (Thu, 27 Mar 2014) | 11 lines
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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r411311 | coreyfarrell | 2014-03-27 15:24:24 -0300 (Thu, 27 Mar 2014) | 11 lines
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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r411315 | coreyfarrell | 2014-03-27 16:15:35 -0300 (Thu, 27 Mar 2014) | 12 lines
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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r411374 | mjordan | 2014-03-28 00:54:31 -0300 (Fri, 28 Mar 2014) | 18 lines
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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r411426 | mjordan | 2014-03-28 01:33:07 -0300 (Fri, 28 Mar 2014) | 25 lines
Blocked revisions 411408
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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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r411442 | mjordan | 2014-03-28 11:18:32 -0300 (Fri, 28 Mar 2014) | 8 lines
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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r411459 | mjordan | 2014-03-28 12:47:36 -0300 (Fri, 28 Mar 2014) | 8 lines
UPGRADE: Note IAX2 compatibility issue between 1.4 and 1.8+ systems.
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r411461 | mjordan | 2014-03-28 13:00:46 -0300 (Fri, 28 Mar 2014) | 2 lines
Remove block on 411408
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r411465 | sgriepentrog | 2014-03-28 13:17:52 -0300 (Fri, 28 Mar 2014) | 31 lines
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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r411512 | mjordan | 2014-03-28 13:48:32 -0300 (Fri, 28 Mar 2014) | 25 lines
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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r411514 | mmichelson | 2014-03-28 14:08:52 -0300 (Fri, 28 Mar 2014) | 3 lines
Add alembic script that adds contact user_agent and endpoint message_context.
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r411529 | mjordan | 2014-03-28 14:35:48 -0300 (Fri, 28 Mar 2014) | 10 lines
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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r411532 | may | 2014-03-28 14:52:09 -0300 (Fri, 28 Mar 2014) | 16 lines
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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r411534 | mjordan | 2014-03-28 15:09:03 -0300 (Fri, 28 Mar 2014) | 22 lines
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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r411586 | file | 2014-04-01 13:51:05 -0300 (Tue, 01 Apr 2014) | 16 lines
app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected.
This patch causes realtime queue members to remain in queues during the reload process. Previously these
members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY".
ASTERISK-23547 #close
ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409)
Review: https://reviewboard.asterisk.org/r/3404/
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r411634 | coreyfarrell | 2014-04-01 17:48:36 -0300 (Tue, 01 Apr 2014) | 14 lines
Blocked revisions 411633
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app_voicemail: fix missing symbol
ASTERISK-23391 caused a regression where the symbol 'defaultlanguage'
was used by app_voicemail but not exported by main/asterisk. This
change renames the variable to ast_defaultlanguage. The variable was
already renamed in Asterisk 12+.
(closes issue ASTERISK-23559)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3408/
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r411636 | rmudgett | 2014-04-01 19:24:39 -0300 (Tue, 01 Apr 2014) | 1 line
stasis_channels.c: Eliminate another overuse of RAII_VAR().
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r411638 | rmudgett | 2014-04-01 19:41:16 -0300 (Tue, 01 Apr 2014) | 11 lines
res_parking: Minor tweaks.
* Use ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of
ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables.
* Use ast_copy_string() instead of inlining it.
* Remove an already done TODO comment.
* Some whitespace tweaks.
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r411668 | mjordan | 2014-04-03 08:43:24 -0300 (Thu, 03 Apr 2014) | 7 lines
res_hep: Fix crash when hep.conf not available
Parts of res_hep properly checked for a valid configuration object before
attempting to access the configuration. A check, however, was missed when
a packet is sent. This patch fixes the crash caused by not checking if the
configuration object is valid.
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r411670 | kmoore | 2014-04-03 08:57:10 -0300 (Thu, 03 Apr 2014) | 7 lines
res_pjsip_pubsub: Add test event for state change
This adds a test event when subscription state changes so that
integration tests may trigger new actions at the appropriate times.
Review: https://reviewboard.asterisk.org/r/3383/
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r411687 | mjordan | 2014-04-04 12:11:48 -0300 (Fri, 04 Apr 2014) | 11 lines
http: Fix spurious ERROR message in responses with no content
When a response has a content length of 0, fwrite would be called to write a
buffer with no data in it. This resulted in the following classic error
message:
[Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success
This patch makes it so that we only attempt to write out the content if the
calculated content_length is non-zero.
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r411701 | rmudgett | 2014-04-04 14:53:20 -0300 (Fri, 04 Apr 2014) | 18 lines
Add some asserts that were handy when looking for a stasis cache problem.
* Assert if a channel is destroyed but has the snapshot staging flag set.
In this case the final channel destruction snapshot would never get taken.
* Assert if what we just got out of the stasis cache is not what we were
looking for. This assert would have saved several days searching for a
bug and a lot of my hair.
* Assert if the music on hold message posts could not find the associated
channel. A crash will happen later when manager tries to send the MOH AMI
message. This assert catches the problem when the stasis message is
posted instead of by the thread processing the defective message.
* Always generate a backtrace when an ast_assert() fails.
Review: https://reviewboard.asterisk.org/r/3411/
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r411717 | rmudgett | 2014-04-04 16:02:57 -0300 (Fri, 04 Apr 2014) | 37 lines
internal_timing: Remove the option and always make it enabled if a timing module is loaded.
The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross. Local channel optimization requires frames
flowing to trigger when optimization can happen. When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing. If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received. With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.
* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed. Asterisk now always uses internal
timing when needed if any timing module is loaded. The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used. The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.
* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().
ASTERISK-22846 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3414/
........
Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11
........
r411790 | kmoore | 2014-04-07 10:24:09 -0300 (Mon, 07 Apr 2014) | 6 lines
PJSIP: Fix crash introduced in r411671
The test event introduced in revision 411671 uses a dangling pointer to
access information about pubsub state changes. This moves the event to
within the lifetime of the pointer.
........
r411804 | kmoore | 2014-04-07 11:28:41 -0300 (Mon, 07 Apr 2014) | 7 lines
Stasis: Fix Stasis() bridge refcount issue
The Stasis() dialplan application monitors what bridge a channel is in
and so necessarily holds on to a bridge pointer. This change ensures
that it also holds on to a reference for that bridge to prevent the
bridge pointer from becoming a dangling pointer.
........
r411809 | wdoekes | 2014-04-07 11:50:04 -0300 (Mon, 07 Apr 2014) | 8 lines
configs: Clean up long line and typo in res_odbc.conf.sample.
........
Merged revisions 411807 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411808 from http://svn.asterisk.org/svn/asterisk/branches/11
........
r411811 | wdoekes | 2014-04-07 11:55:19 -0300 (Mon, 07 Apr 2014) | 5 lines
app_queue: Re-add HoldTime to QueueCallerAbandon event (simple typo during ast12 refactor).
Reported by: Ibrahim22 (on IRC)
Tested by: Ibrahim22
........
Merged revisions 410830,410844,410858,410861,410876,410890,410904,410918,410933,410949,410966,410995,411023,411086,411091,411141,411157,411159,411173,411191,411193,411245,411295,411311,411315,411374,411426,411442,411459,411461,411465,411512,411514,411529,411532,411534,411586,411634,411636,411638,411668,411670,411687,411701,411717,411790,411804,411809,411811 from http://svn.asterisk.org/svn/asterisk/branches/12
Added:
team/file/bridge_unreal_optimizer/configs/hep.conf.sample
- copied unchanged from r411811, branches/12/configs/hep.conf.sample
team/file/bridge_unreal_optimizer/contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py
- copied unchanged from r411811, branches/12/contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py
team/file/bridge_unreal_optimizer/include/asterisk/res_hep.h
- copied unchanged from r411811, branches/12/include/asterisk/res_hep.h
team/file/bridge_unreal_optimizer/res/res_hep.c
- copied unchanged from r411811, branches/12/res/res_hep.c
team/file/bridge_unreal_optimizer/res/res_hep.exports.in
- copied unchanged from r411811, branches/12/res/res_hep.exports.in
team/file/bridge_unreal_optimizer/res/res_hep_pjsip.c
- copied unchanged from r411811, branches/12/res/res_hep_pjsip.c
Removed:
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/iaxfriends.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/meetme.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/musiconhold.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/queue_log.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/sippeers.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/voicemail.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/voicemail_data.sql
team/file/bridge_unreal_optimizer/contrib/realtime/mysql/voicemail_messages.sql
team/file/bridge_unreal_optimizer/contrib/realtime/postgresql/realtime.sql
Modified:
team/file/bridge_unreal_optimizer/ (props changed)
team/file/bridge_unreal_optimizer/CHANGES
team/file/bridge_unreal_optimizer/UPGRADE.txt
team/file/bridge_unreal_optimizer/addons/chan_ooh323.c
team/file/bridge_unreal_optimizer/addons/ooh323c/src/ooCmdChannel.c
team/file/bridge_unreal_optimizer/addons/ooh323c/src/ooGkClient.c
team/file/bridge_unreal_optimizer/addons/ooh323c/src/oochannels.c
team/file/bridge_unreal_optimizer/addons/ooh323c/src/ooh323.c
team/file/bridge_unreal_optimizer/addons/ooh323c/src/ooq931.c
team/file/bridge_unreal_optimizer/apps/app_confbridge.c
team/file/bridge_unreal_optimizer/apps/app_jack.c
team/file/bridge_unreal_optimizer/apps/app_queue.c
team/file/bridge_unreal_optimizer/apps/app_speech_utils.c
team/file/bridge_unreal_optimizer/apps/app_stack.c
team/file/bridge_unreal_optimizer/apps/app_voicemail.c
team/file/bridge_unreal_optimizer/apps/confbridge/conf_config_parser.c
team/file/bridge_unreal_optimizer/channels/chan_iax2.c
team/file/bridge_unreal_optimizer/channels/chan_sip.c
team/file/bridge_unreal_optimizer/channels/pjsip/dialplan_functions.c
team/file/bridge_unreal_optimizer/channels/sip/include/sip.h
team/file/bridge_unreal_optimizer/configs/asterisk.conf.sample
team/file/bridge_unreal_optimizer/configs/res_odbc.conf.sample
team/file/bridge_unreal_optimizer/configs/sip.conf.sample
team/file/bridge_unreal_optimizer/configs/sorcery.conf.sample
team/file/bridge_unreal_optimizer/contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py
team/file/bridge_unreal_optimizer/funcs/func_blacklist.c
team/file/bridge_unreal_optimizer/funcs/func_callcompletion.c
team/file/bridge_unreal_optimizer/funcs/func_callerid.c
team/file/bridge_unreal_optimizer/funcs/func_cdr.c
team/file/bridge_unreal_optimizer/funcs/func_channel.c
team/file/bridge_unreal_optimizer/funcs/func_dialplan.c
team/file/bridge_unreal_optimizer/funcs/func_frame_trace.c
team/file/bridge_unreal_optimizer/funcs/func_global.c
team/file/bridge_unreal_optimizer/funcs/func_groupcount.c
team/file/bridge_unreal_optimizer/funcs/func_jitterbuffer.c
team/file/bridge_unreal_optimizer/funcs/func_math.c
team/file/bridge_unreal_optimizer/funcs/func_odbc.c
team/file/bridge_unreal_optimizer/funcs/func_pitchshift.c
team/file/bridge_unreal_optimizer/funcs/func_speex.c
team/file/bridge_unreal_optimizer/funcs/func_strings.c
team/file/bridge_unreal_optimizer/funcs/func_volume.c
team/file/bridge_unreal_optimizer/include/asterisk.h
team/file/bridge_unreal_optimizer/include/asterisk/channel.h
team/file/bridge_unreal_optimizer/include/asterisk/manager.h
team/file/bridge_unreal_optimizer/include/asterisk/options.h
team/file/bridge_unreal_optimizer/include/asterisk/res_odbc.h
team/file/bridge_unreal_optimizer/include/asterisk/res_pjsip.h
team/file/bridge_unreal_optimizer/include/asterisk/rtp_engine.h
team/file/bridge_unreal_optimizer/include/asterisk/sorcery.h
team/file/bridge_unreal_optimizer/main/asterisk.c
team/file/bridge_unreal_optimizer/main/bucket.c
team/file/bridge_unreal_optimizer/main/cdr.c
team/file/bridge_unreal_optimizer/main/channel.c
team/file/bridge_unreal_optimizer/main/channel_internal_api.c
team/file/bridge_unreal_optimizer/main/core_unreal.c
team/file/bridge_unreal_optimizer/main/dial.c
team/file/bridge_unreal_optimizer/main/features_config.c
team/file/bridge_unreal_optimizer/main/format.c
team/file/bridge_unreal_optimizer/main/http.c
team/file/bridge_unreal_optimizer/main/manager.c
team/file/bridge_unreal_optimizer/main/message.c
team/file/bridge_unreal_optimizer/main/say.c
team/file/bridge_unreal_optimizer/main/sorcery.c
team/file/bridge_unreal_optimizer/main/stasis_cache.c
team/file/bridge_unreal_optimizer/main/stasis_channels.c
team/file/bridge_unreal_optimizer/main/tcptls.c
team/file/bridge_unreal_optimizer/main/utils.c
team/file/bridge_unreal_optimizer/res/ari/config.c
team/file/bridge_unreal_optimizer/res/ari/resource_bridges.h
team/file/bridge_unreal_optimizer/res/ari/resource_channels.c
team/file/bridge_unreal_optimizer/res/parking/parking_bridge.c
team/file/bridge_unreal_optimizer/res/res_ari.c
team/file/bridge_unreal_optimizer/res/res_calendar.c
team/file/bridge_unreal_optimizer/res/res_config_odbc.c
team/file/bridge_unreal_optimizer/res/res_fax_spandsp.c
team/file/bridge_unreal_optimizer/res/res_jabber.c
team/file/bridge_unreal_optimizer/res/res_musiconhold.c
team/file/bridge_unreal_optimizer/res/res_mutestream.c
team/file/bridge_unreal_optimizer/res/res_mwi_external.c
team/file/bridge_unreal_optimizer/res/res_odbc.c
team/file/bridge_unreal_optimizer/res/res_odbc.exports.in
team/file/bridge_unreal_optimizer/res/res_pjsip.c
team/file/bridge_unreal_optimizer/res/res_pjsip/config_system.c
team/file/bridge_unreal_optimizer/res/res_pjsip/pjsip_configuration.c
team/file/bridge_unreal_optimizer/res/res_pjsip/pjsip_options.c
team/file/bridge_unreal_optimizer/res/res_pjsip_endpoint_identifier_ip.c
team/file/bridge_unreal_optimizer/res/res_pjsip_header_funcs.c
team/file/bridge_unreal_optimizer/res/res_pjsip_messaging.c
team/file/bridge_unreal_optimizer/res/res_pjsip_pubsub.c
team/file/bridge_unreal_optimizer/res/res_pjsip_registrar.c
team/file/bridge_unreal_optimizer/res/res_stasis.c
team/file/bridge_unreal_optimizer/res/res_stasis_snoop.c
team/file/bridge_unreal_optimizer/res/res_xmpp.c
team/file/bridge_unreal_optimizer/rest-api/api-docs/applications.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/asterisk.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/bridges.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/channels.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/deviceStates.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/endpoints.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/events.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/mailboxes.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/playbacks.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/recordings.json
team/file/bridge_unreal_optimizer/rest-api/api-docs/sounds.json
team/file/bridge_unreal_optimizer/rest-api/resources.json
team/file/bridge_unreal_optimizer/tests/test_sorcery.c
team/file/bridge_unreal_optimizer/tests/test_sorcery_astdb.c
team/file/bridge_unreal_optimizer/tests/test_sorcery_realtime.c
Propchange: team/file/bridge_unreal_optimizer/
------------------------------------------------------------------------------
--- branch-11-blocked (original)
+++ branch-11-blocked Mon Apr 7 11:05:17 2014
@@ -1,1 +1,1 @@
-/branches/11:373240,375247,375702,385356,395020,396441,397034,397604,399305,401379,405362,405380,407457,409002,409208,409524
+/branches/11:373240,375247,375702,385356,395020,396441,397034,397604,399305,401379,405362,405380,407457,409002,409208,409524,411633
Propchange: team/file/bridge_unreal_optimizer/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/file/bridge_unreal_optimizer/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Apr 7 11:05:17 2014
@@ -1,1 +1,1 @@
-/branches/12:1-410827
+/branches/12:1-411867
Modified: team/file/bridge_unreal_optimizer/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/file/bridge_unreal_optimizer/CHANGES?view=diff&rev=411869&r1=411868&r2=411869
==============================================================================
--- team/file/bridge_unreal_optimizer/CHANGES (original)
+++ team/file/bridge_unreal_optimizer/CHANGES Mon Apr 7 11:05:17 2014
@@ -22,6 +22,10 @@
* The live recording object on recording events now contains a target_uri
field which contains the URI of what is being recorded.
+ * The bridge type used when creating a bridge is now a comma separated list of
+ bridge properties. Valid options are: mixing, holding, dtmf_events, and
+ proxy_media.
+
* A channelId can now be provided when creating a channel, either in the
uri (POST channels/my-channel-id) or as query parameter. A local channel
will suffix the second channel id with ';2' unless provided as query
@@ -42,10 +46,58 @@
second channel when dialing LOCAL, or defaults to appending ;2 if only
the single Id is given.
+RealTime
+------------------
+ * A new set of Alembic scripts has been added for CDR tables. This will create
+ a 'cdr' table with the default schema that Asterisk expects.
+
+res_hep
+------------------
+ * A new module, res_hep, has been added, that acts as a generic packet
+ capture agent for the Homer Encapsulation Protocol (HEP) version 3.
[... 4130 lines stripped ...]
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