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r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines
Merged revisions 298597 via svnmerge from
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r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines
Merged revisions 298596 via svnmerge from
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r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would
not try and hang up the local channel it created, which would kill the call.
ABE-2624
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r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) | 8 lines
Patch for deadlock from ordering issue between channel/queue locks in app_queue
(set_queue_variables).
(closes issue #18031)
Reported by: rain
Review: https://reviewboard.asterisk.org/r/1018/
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r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines
Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.
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r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
Simplify locking code for REDIRECTING interception macro when forwarding a call.
Simplified the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
for launching the REDIRECTING interception macro when a call is forwarded.
Reduced the lock time of the 'o->chan' and 'in' channels.
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r287388 | tilghman | 2010-09-17 16:08:54 -0500 (Fri, 17 Sep 2010) | 21 lines
Merged revisions 287387 via svnmerge from
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r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines
Merged revisions 287386 via svnmerge from
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r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines
Blank columns should get set on reload, not ignored.
(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
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r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines
Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf.
Review: https://reviewboard.asterisk.org/r/922/
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r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines
Merged revisions 284631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
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r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707)
Reported by: ira
Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/876/
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r280161 | seanbright | 2010-07-28 12:52:12 -0400 (Wed, 28 Jul 2010) | 15 lines
Merged revisions 280160 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines
Plug a reference leak in app_queue when adding members dynamically.
(closes issue #17738)
Reported by: bobwienholt
Patches:
issue17738.patch uploaded by bobwienholt (license 950)
Tested by: bobwienholt, seanbright
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r279227 | rmudgett | 2010-07-23 17:20:47 -0500 (Fri, 23 Jul 2010) | 21 lines
Merged revisions 279207 via svnmerge from
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r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines
Merged revisions 279206 via svnmerge from
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r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines
SIP promiscuous redirect could fail to dial the redirect.
The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
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Just say the number of seconds (after minutes) rather than doing some incorrect
calculation with respect to minutes.
(closes issue #17498)
Reported by: corruptor
Patches:
holdesecs_bug.diff uploaded by corruptor (license 253)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17471)
Reported by: jazzy
Patches:
app_queue.c.diff uploaded by jazzy (license 1056)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Review: https://reviewboard.asterisk.org/r/777/
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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force directly to the default case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers.
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r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines
Send AgentComplete manager events in the event of blind and attended transfers.
(closes issue #16819)
Reported by: elbriga
Patches:
app_queue.diff uploaded by elbriga (license 482)
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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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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Closes AST-33)
Reviewboard: https://reviewboard.asterisk.org/r/368/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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