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Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines
Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670.
(closes issue #17334)
Reported by: jvandal
Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
Tested by: aragon, jvandal
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r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines
Don't hang up on a queue caller if the file we attempt to play does not exist.
This also fixes a documentation mistake in file.h that made my original attempt
to correct this problem not work correctly.
(closes issue #17061)
Reported by: RoadKill
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The connected line update macro would not get run if the connected line
number string was empty. The number could be empty if the connected line
update did not update a number but the name. It should be run if there
was an AST_CONTROL_CONNECTED_LINE frame received for pending dials and
queues.
Renamed and added some more comments for some confusing identifiers
directly connected to the related code.
Also fixed a memory leak in app_queue.
Review: https://reviewboard.asterisk.org/r/669/
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From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
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Resets each member's lastcall to 0 now.
(closes issue #17262)
Reported by: rain
Patches:
wrapuptime_reset_fix.diff uploaded by rain (license 327)
Tested by: rain
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in all queues.
See the CHANGES file and queues.conf.sample for more details.
(closes issue #17008)
Reported by: jlpedrosa
Patches:
queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002)
Review: https://reviewboard.asterisk.org/r/581/
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When a specific position is specified for the queue, the idea
was that the caller cannot be placed ahead of higher-priority
callers. Unfortunately, the logic was reversed so that the caller
could ONLY be placed ahead of higher priority callers.
Discovered while writing a unit test.
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This module implements an abstraction for retrieving and exporting
asterisk data.
Developed by:
Brett Bryant <brettbryant@gmail.com>
Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h
Review: https://reviewboard.asterisk.org/r/275/
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SWP-1229
ABE-2161
* Ensure chan_local.c:local_call() will not leak cid.cid_dnid when
copying.
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(closes issue #15494)
Reported by: makoto
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Only chan_dahdi set a value in cdrflags. Everyone else just copied it
around the system. Noone cared about any value it may have contained.
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(closes issue #16475)
Reported by: okrief
Patches:
queue_crash.diff uploaded by dvossel (license 671)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines
Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls.
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r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines
fixes random deadlock in app_queue with use_weight during reload
(closes issue #16677)
Reported by: tim_ringenbach
Patches:
app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540)
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r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) | 5 lines
Revert 243570, I should have looked at this closer. Will reopen the issue, but
am leaving the review closed as the change was pointless.
(issue #16488)
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r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) | 9 lines
Extend announcement URL used with Queue from 80 chars to PATH_MAX.
(closes issue #16488)
Reported by: syspert
Patches:
soundfilelen.pacth-2 uploaded by syspert (license 938)
Review: https://reviewboard.asterisk.org/r/475/
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(closes issue #16078)
Reported by: RoadKill
Patches:
quequename_limit.patch uploaded by ppyy (license 906)
Tested by: dvossel
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When reporting hold time, the number of seconds should be mod 60.
Otherwise audio playback could be something like "2 minutes 123 seconds"
rather than "2 minutes 3 seconds".
Also, the "minute" sound file is missing, so for the moment until
that file can be created the "minutes" file is used instead.
(closes issue #16168)
Reported by: nickilo
Patches:
patch-unified-trunk-rev-222176 uploaded by nickilo (license )
Tested by: nickilo, wonderg
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16385)
Reported by: haakon
Patches:
app_queue.c.patch uploaded by haakon (license 880)
app_queue.c.patch_v2 uploaded by dvossel (license 671)
Tested by: haakon
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The QUEUE_MEMBER dialplan function can return total members,
logged-in members and "free" members count. A member is counted
as "free" immediately after his call ends, even though its wrap-up
time, if specified in queues.conf, has not yet expired, and the
queue will not actually route a call to it.
This Patch introduces a new "ready" option that only counts
free agents no longer in the wrap up time period.
(closes issue #16240)
Reported by: kkm
Patches:
appqueue-memberfun-readyoption-trunk.diff uploaded by kkm (license 888)
Tested by: kkm, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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to the feature patch
(issue #16384)
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The 'R' argument stops moh and indicates ringing once the agent is
ringing. This allows the person in the queue to know their call
is potentially about to be answered.
(closes issue #16384)
Reported by: haakon
Patches:
new_app_queue.c.patch uploaded by haakon (license 880)
Tested by: haakon, loloski, dvossel
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members when enabled
(closes issue #14559)
Reported by: fiddur
Patches:
trunk-199584-1.diff uploaded by fiddur (license 678)
Tested by: fiddur, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds a ref to the queue_ent object's parent call_queue
in queue_exec() so the call_queue won't be destroyed
while the the queue_ent still holds a pointer to it.
(closes issue 0015686)
Tested by: dvossel, aragon
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debug statements.
(closes issue #15982, closes issue #15984)
Reported by: atis
Patches:
20091111__issue15982.diff.txt uploaded by tilghman (license 14)
Tested by: atis
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(Closes AST-33)
Reviewboard: https://reviewboard.asterisk.org/r/368/
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calculating the time of announcments from the end of the previous announcment rather than from the beginning.
(closes issue #15260)
Reported by: tonils
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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format hint:<extension>@<context>.
(closes issue #15168)
Reported by: p_lindheimer
Patches:
queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines
Don't ring another channel, if there's not enough time for a queue member to answer.
(Fixes AST-228)
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(closes issue #15146)
Reported by: arabe
Patches:
asterisk-trunk.patch uploaded by arabe (license 786)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug 2009) | 10 lines
This patch adds additional checking when generating queue log TRANSFER events.
The additional checks prevent generation of false TRANSFER events in certain situations.
(closes issue #14536)
Reported by: aragon
Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson
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r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines
QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername.
This is a partial revert of revision 82590, which was an attempted cleanup,
but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended
as a method by which component interfaces could be queried from the queue.
Membername isn't useful here, because that field cannot be used to obtain
further information about the member. See the documentation on
QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various
AMI commands which take a member argument for further justification.
(closes issue #15664)
Reported by: rain
Patches:
app_queue-queue_member_list.diff uploaded by rain (license 327)
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r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
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Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.
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This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
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extension from a queue.
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When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
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