Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines
Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182945 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines
Change faulty comparison used when announcing average hold minutes and seconds
(closes issue #14227)
Reported by: caspy
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182122 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines
Run the macro on the queue member's channel when he answers, not the caller's channel.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181848 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines
Fix crash when sleep and retries argument was not given to RetryDial application.
(closes issue #14647)
Reported by: sherpya
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181613 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines
Merged revisions 180567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines
Make compilation succeed in dev-mode when IMAP storage is enabled.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180582 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines
Merged revisions 180464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines
[IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts.
There was a fix put in a while back so that an X-Asterisk-VM-Context message header was
added to stored IMAP voicemails. This would allow for us to differentiate if the same
mailbox name was used in multiple contexts. The problem still left was that not all places
where messages were retrieved actually attempted to use this header for information when
retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain
work as expected.
(closes issue #13853)
Reported by: vicks1
Patches:
13853_v2.patch uploaded by mmichelson (license 60)
Tested by: lmadsen
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180466 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines
Merged revisions 180380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines
Fix broken mailbox parsing when searchcontexts option is enabled.
When using the searchcontexts option in voicemail.conf, the code
made the assumption that all mailbox names defined were unique across
all contexts. However, the code did nothing to actually enforce this
assumption, nor did it do anything to alert a user that he may have
created an ambiguity in his voicemail.conf file by defining the same
mailbox name in multiple contexts.
With this change, we now will issue a nice long warning if searchcontexts
is on and we encounter the same mailbox name in multiple contexts and ignore
any duplicates after the first box. Whether searchcontexts is enabled or not,
if we come across a duplicate mailbox in the same context, then we will issue
a warning and ignore the duplicated mailbox. I have also added a small note
to voicemail.conf.sample in the explanation for searchcontexts explaining
that you cannot define the same mailbox in multiple contexts if you have
enabled the option.
(closes issue #14599)
Reported by: lmadsen
Patches:
14599.patch uploaded by mmichelson (license 60) (with slight modification)
Tested by: lmadsen
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180404 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
Remove duplicate 'k' and 'K' Dial options.
(closes issue #14601)
Reported by: alecdavis
Patches:
app_dial.optionk.diff.txt uploaded by alecdavis (license 585)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180121 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
app_read does not break from prompt loop with user terminated empty string
In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata().
(closes issue #14279)
Reported by: Marquis
Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671)
Tested by: Marquis, dvossel
Review: http://reviewboard.digium.com/r/177/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180078 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines
Merged revisions 180006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank"
value for a sound file in queues.conf and have no sound played
back. The problem with this is that it would result in some ugly
CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in queues.conf.sample
(closes issue #14227)
Reported by: caspy
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180008 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This was discovered while looking at issue #14353
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179971 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines
Merged revisions 179532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines
Move ast_waitfor() down to avoid the results of the API call becoming stale.
This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice. By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.
So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available. Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.
This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk. He was using the timerfd timing module. When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was
the cause of the last legitimate call to ast_read() done by autoservice.
In this test, an IAX2 channel was calling into the MeetMe conference. It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled. Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled. So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.
Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function. The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused Asterisk
to lock up very quickly.
Thanks to dvossel and mmichelson for the fun debugging session. :-)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179534 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
use it.
(closes issue #14571)
Reported by: zktech
Patches:
20090302__bug14571.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179363 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to
this commit.
(closes issue #14338)
Reported by: fiddur
Patches:
14338.patch uploaded by mmichelson (license 60)
Tested by: fiddur
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179222 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
If BIT_TYPES_DEFINED gets defined before linux/types.h is included, the
__s32 type doesn't get defined
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178306 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r177699 | dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
Make app_fax compatible with spandsp-0.0.6pre4
Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
integer to indicate the number of pages transferred (so far) during the fax
session. The spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and pages_rx. This
revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
compatibility for previous spandsp releases.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177763 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r148868 | tilghman | 2008-10-14 10:15:45 -0500 (Tue, 14 Oct 2008) | 6 lines
API differences in spandsp 0.0.6pre1 and higher
(closes issue #13688)
Reported by: irroot
Patches:
app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177762 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines
Allow semicolons to be escaped, when passing arguments to the System command.
(closes issue #14231)
Reported by: jcovert
Patches:
20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551)
Tested by: jcovert
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177665 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) | 2 lines
Oops, merge broke trunk
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177662 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r177537 | tilghman | 2009-02-19 16:33:00 -0600 (Thu, 19 Feb 2009) | 14 lines
Merged revisions 177536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines
Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads.
(closes issue #14253)
Reported by: Skavin
Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177538 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) | 10 lines
Merged revisions 177383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines
If we are able to create a speech structure unset the ERROR variable in case it was previously set.
(issue #LUMENVOX-13)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177385 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r177101 | russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines
Re-add 'o' option to MeetMe, reverting rev 62297.
Enabling this option by default proved to be a bad idea, as the talker detection
is not very reliable. So, make it optional again, and off by default.
(issue #13801)
Reported by: justdave
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177135 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) | 24 lines
Merged revisions 176249,176252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines
Open the DAHDI pseudo device and set it to be nonblocking atomically
Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately
from opening the file was causing an "inappropriate ioctl for device" error.
While I cannot fathom why this would be happening, I certainly am not opposed
to making the code a bit more compact/efficient if it also fixes a bug.
(closes issue #14482)
Reported by: ys
Patches:
meetme.patch uploaded by ys (license 281)
Tested by: ys
........
r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines
Remove unused variable and make dev-mode compilation happy
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176256 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines
Add an option to keep the recorded file upon hangup.
(closes issue #14341)
Reported by: fnordian
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175550 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 35 lines
Fix odd "thank you" sound playing behavior in app_queue.c
If someone has configured the queue to play an position or holdtime
announcement, then it is odd and potentially unexpected to hear a
"Thank you for your patience" sound when no position or holdtime
was actually announced.
This fixes the announcement so that the "thanks" sound is only played
in the case that a position or holdtime was actually announced.
There is a way that the "thank you" sound can be played without a
position or holdtime, and that is to set announce-frequency to a value
but keep announce-position and announce-holdtime both turned off.
(closes issue #14227)
Reported by: caspy
Patches:
14227_v3.patch uploaded by putnopvut (license 60)
Tested by: caspy
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174949 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines
Fix 'd' option for app_dial and add new option to Answer application
The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.
I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.
I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.
I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.
All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches
(closes issue #14164)
Reported by: DennisD
Patches:
14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/145
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174946 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r174805 | mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 lines
Fix potential for stack overflows in app_chanspy.c
When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.
The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174820 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
this fix. I was wrong. Here it is.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174439 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines
Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage
(closes issue #13905)
Reported by: jaroth
Patches:
foldermove_v2.patch uploaded by jaroth (license 50)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173774 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines
Merged revisions 173696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
Add new configuration option to make shared IMAP mailboxes function as expected.
The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
using the same IMAP storage location to function as one mailbox. This allows
all messages to be retrieved for any user in the group. The patch alters the
'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
for a given user.
(closes issue #13673)
Reported by: howardwilkinson
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173698 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines
Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines
Fix situations where queue members could be autopaused unexpectedly
Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.
(closes issue #14376)
Reported by: fiddur
Patches:
14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173694 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines
Merged revisions 173592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines
Add some missing cleanup to app_mixmonitor
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173594 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines
Merged revisions 173559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines
Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).
The solution for this is to employ a datastore, which has the nice benefit of allowing us
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!
(closes issue #14374)
Reported by: aragon
Patches:
14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173590 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines
Fix some areas where the incorrect interface was passed to ast_device_state
I swear it feels like I already did this once...
(closes issue #14359)
Reported by: francesco_r
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173534 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines
Merged revisions 173396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines
Revert my previous change because it was stupid
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173398 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines
Merged revisions 173392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines
Add a missing unlock. Extremely unlikely to ever matter, but it's needed.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173394 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines
Blank argument crashes Asterisk
(closes issue #14377)
Reported by: amorsen
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172742 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172635 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines
Merged revisions 172438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172503 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines
Merged revisions 172030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172065 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines
Fix queue crashes that would occur after the calling channel was masqueraded.
The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.
To fix this problem, there are two measures taken
1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c
(closes issue #14260)
Reported by: ccesario
Patches:
14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171619 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines
Merged revisions 170979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines
Resolve a logic error that was causing Page() to crash when more than one
channel was specified.
(closes issue #14308)
Reported by: bluefox
Patches:
20090124__bug14308.diff.txt uploaded by seanbright (license 71)
Tested by: kc0bvu
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170981 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines
Merged revisions 170568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines
When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.
(closes issue #14310)
Reported by: RadicAlish
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170570 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines
Merged revisions 170147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282)
Reported by: cheesegrits
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170149 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines
Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop.
(closes issue #14304)
Reported by: jcovert
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170048 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines
Fix device state parsing issues for channel names with multiple slashes
The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.
In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.
(closes issue #14014)
Reported by: kebl0155
Patches:
14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@169612 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r169365 | tilghman | 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) | 11 lines
Merged revisions 169364 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines
Truncate userevents at the end of a line, when the command exceeds the buffer.
(closes issue #14278)
Reported by: fnordian
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@169366 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines
Merged revisions 168828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines
Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475)
Reported by: chappell
Patches:
vm_multilang.patch uploaded by chappell (license 8)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@168835 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines
Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
(closes issue #14117)
Reported by: sergedevorop
Patches:
20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
Tested by: sergedevorop
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@168706 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines
Merged revisions 168628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
(closes issue #14086)
Reported by: ZX81
Patches:
14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@168630 f38db490-d61c-443f-a65b-d21fe96a405b
|