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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines
(closes issue #13251)
Reported by: sergee
Tested by: murf
THis is a bold move for a static release fix, but I wouldn't have
made it if I didn't feel confident (at least a *bit* confident)
that it wouldn't mess everyone up.
The reasoning goes something like this:
1. We simply cannot do anything with CDR's at the current point
(in pbx.c, after the __ast_pbx_run loop). It's way too late to
have any affect on the CDRs. The CDR is already posted and gone,
and the remnants have been cleared.
2. I was very much afraid that moving the running of the 'h'
extension down into the bridge code (where it would be now
practical to do it), would result in a lot more calls to the
'h' exten, so I implemented it as another exten under another
name, but found, to my pleasant surprise, that there was a
1:1 correspondence to the running of the 'h' exten in the
pbx_run loop, and the new spot at the end of the bridge.
So, I ifdef'd out the current 'h' loop, and moved it into
the bridge code. The only difference I can see is the stuff
about the AST_PBX_KEEPALIVE, and hopefully, if this
is still an important decision point, I can replicate it
if there are complaints. To be perfectly honest,
the KEEPALIVE situation is not totally clear to me,
and how it relates to a post-bridge situation is less
clear. I suspect the users will point out everything
in total clarity if this steps on anyone's toes!
3. I temporarily swap the bridge_cdr into the channel
before running the 'h' exten, which makes it possible
for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set,
the users can also get at the billsec/duration vals.
After the h exten finishes, the cdr is swapped back
and processing continues as normal.
Please, all who deal with CDR's, please test this version
of Asterisk, and file bug reports as appropriate!
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I also made a little fix to the app_dial's 'e' option,
that is related to my updates.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines
Fix a crash in the ChanSpy application. The issue here is that if you call
ChanSpy and specify a spy group, and sit in the application long enough looping
through the channel list, you will eventually run out of stack space and the
application with exit with a seg fault. The backtrace was always inside of
a harmless snprintf() call, so it was tricky to track down. However, it turned
out that the call to snprintf() was just the biggest stack consumer in this
code path, so it would always be the first one to hit the boundary.
(closes issue #13338)
Reported by: ruddy
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug 2008) | 23 lines
Add a lock and unlock prior to the destruction of the chanspy_ds
lock to ensure that no other threads still have it locked. While
this should not happen under normal circumstances, it appears that
if the spyer and spyee hang up at nearly the same time, the following
may occur.
1. ast_channel_free is called on the spyee's channel.
2. The chanspy datastore is removed from the spyee's channel in
ast_channel_free.
3. In the spyer's thread, the spyer attempts to remove and destroy the datastore
from the spyee channel, but the datastore has already been removed in step 2,
so the spyer continues in the code.
4. The spyee's thread continues and calls the datastore's destroy callback,
chanspy_ds_destroy. This involves locking the chanspy_ds.
5. Now the spyer attempts to destroy the chanspy_ds lock. The problem is that in step 4,
the spyee has locked this lock, meaning that the spyer is attempting to destroy a lock
which is currently locked by another thread.
The backtrace provided in issue #12969 supports the idea that this is possible
(and has even occurred). This commit does not close the issue, but should help
in preventing one type of crash associated with the use of app_chanspy.
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and confusing code pieces. Clarify the logic within
queues.conf.sample.
(closes issue #12690)
Reported by: atis
Patches:
queue_timeoutpriority.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines
Change the inequalities used in app_queue with regards
to timeouts from being strict to non-strict for more
accuracy.
(closes issue #13239)
Reported by: atis
Patches:
app_queue_timeouts_v2.patch uploaded by atis (license 242)
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(closes issue #13252)
Patches:
v1-13252.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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files from main/
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app_voicemail and app_queue.
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debug messages.
Thanks to eliel for alerting me.
No thanks to buildbot.
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trunk.
For an explanation of what "imap_consistency" is,
please see svn revision 134223 to the 1.4 branch.
Coincidentally, this also fixes a recent bug report
regarding the inability to save messages to the new
folder when using IMAP storage since they will would
be flagged as "seen" and not be recognized as new
messages.
(closes issue #13234)
Reported by: jaroth
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) | 7 lines
Update persistent state on all exit conditions.
(closes issue #12916)
Reported by: sgenyuk
Patches:
app_queue.patch.txt uploaded by neutrino88 (license 297)
Tested by: sgenyuk, aragon
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(closes issue #13058)
Reported by: pputman
Patches:
app_meetme_aststr2.patch uploaded by pputman (license 81)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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channel related, and add the ability to add/find/remove datastores to manager sessions
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) | 6 lines
Memory leak on unload
(closes issue #13231)
Reported by: eliel
Patches:
app_voicemail.leak.patch uploaded by eliel (license 64)
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such as "Work" and "Family" would be subfolders of the
INBOX. This is an invalid assumption to make, but it could
be desirable to set up folders in this manner, so a new
option for voicemail.conf, "imapparentfolder" has been
added to allow for this.
(closes issue #13142)
Reported by: jaroth
Patches:
parentfolder.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) | 2 lines
make app_ices compile on OpenBSD.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) | 9 lines
Get app_ices working again
(closes issue #12981)
Reported by: dlogan
Patches:
20080709__app_ices_v2_update_trunk.diff uploaded by bbryant (license 36)
20080709__app_ices_v2_update_14.diff uploaded by bbryant (license 36)
Tested by: bbryant
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines
Add more timeout checks into app_queue, specifically
targeting areas where an unknown and potentially
long time has just elapsed. Also added a check
to try_calling() to return early if the timeout
has elapsed instead of potentially setting a negative
timeout for the call (thus making it have *no* timeout
at all).
(closes issue #13186)
Reported by: miquel_cabrespina
Patches:
13186.diff uploaded by putnopvut (license 60)
Tested by: miquel_cabrespina
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https://origsvn.digium.com/svn/asterisk/trunk
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r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines
Fix the parsing of the "reason" parameter in the
Diversion: header.
(closes issue #13195)
Reported by: woodsfsg
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UPGRADE-1.4.txt)
(Closes issue #13181)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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features that will always be present in DAHDI
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines
Detect when sox fails to raise the volume, because sox can't read the file.
(closes issue #12939)
Reported by: rickbradley
Patches:
20080728__bug12939.diff.txt uploaded by Corydon76 (license 14)
Tested by: rickbradley
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(closes issue #13134)
Reported by: eliel
Patches:
app_image.c.patch uploaded by eliel (license 64)
UPGRADE.patch uploaded by eliel (license 64)
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(closes issue #13155)
Reported by: greenfieldtech
Patches:
app_voicemail.c.patch uploaded by greenfieldtech (license 369)
hebrew.ods uploaded by greenfieldtech (license 369)
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Pointed out by Atis Lezdins in #asterisk-dev
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
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r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines
Zap/pseudo is ten characters, but DAHDI/pseudo is
twelve. The strncmp call in next_channel should
account for this.
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r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines
Update the "last" channel in next_channel in app_chanspy so
that the same pseudo channel isn't constantly returned.
related to issue #13124
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supported on a channel (yet _another_ useful patch by eliel).
(closes issue #13081)
Reported by: eliel
Patches:
app_sendtext.c.patch uploaded by eliel (license 64)
Tested by: eliel
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own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
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impact on my machine ..
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the SendDTMF application. Also correct the default
pause between digits.
(closes issue #13102)
Reported by: eliel
Patches:
app_senddtmf.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13082)
Reported by: eliel
Patches:
app_rpt.c.patch uploaded by eliel (license 64)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
(closes issue #13084)
Reported by: elbriga
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r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines
Apparently, "thread safety" is important, whatever
that means. :P
(Thanks Russell!)
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r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
(closes issue #13047)
Reported by: festr
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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a member of any queue.
(closes issue #13073)
Reported by: eliel
Patches:
app_queue.c.patch uploaded by eliel (license 64)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
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(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
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