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This deadlock was introduced by the fix to ensure that channels are properly
locked when handling channel variables. There were sections of this code where
the channel pvt was locked before the channel lock, when in fact it _must_ be
the other way around.
(closes issue #11582)
Reported by: bugi
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's' extension.
(closes issue #9099)
Reported by: kb1_kanobe2
Patches:
20070906__9099.diff.txt uploaded by Corydon76 (license 14)
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I wonder if Asterisk can run at all without res_features. My guess is that
there's propably a lot of more modules and the core that depends on it.
Reported by: caio1982
(closes issue #11574)
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Reported by: julianjm
Patch by: julianjm
(Closes issue #10238)
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O_CREAT.
Issue 11238, reported by puzzled.
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(closes issue #11166)
Reported by: eliel
Patches:
chan_zap.c.patch uploaded by eliel (license 64)
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Part of issue 11017, patch by tzafrir.
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Simplify code slightly in the process.
Issue 11079, reported by ddv2005
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at the same time without doing extra work to make sure it succeeds.
(closes issue #10895, patch by me)
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place, so don't duplicate it
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This will end up making the channel variable global, which is not right.
Closes issue #10720, patch by flefoll.
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that the interface list consists of channels 1 through <x> with no gaps in the sequence
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ast_read to return NULL in the case that the channel has been hung up.
(crash reported by anonymouz666 on IRC in #asterisk-dev)
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Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) | 4 lines
Revert patch committed for issue #9660. It broke E&M trunks.
(closes issue #10360)
(closes issue #10364)
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W9SH), and oversight in logic in TONE_VERIFY/RELAX mode in chan_zap.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 line
this fixes bug 10293, where the error message because defaultzone or loadzone was not defined was confusing
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | 12 lines
(closes issue #9660)
Reported by: mmacvicar
Patches submitted by: bbryant, russell
Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous
When using a TDM400P (and probably other analog cards) there was a chance that
you could hang up and pick the phone back up where it has been long enough to
be not considered a flash hook, but too soon such that the device reports that
it is busy and the person on the phone will only hear silence. This patch
makes chan_zap more tolerant of this and gives the device a couple of seconds
to succeed so the person on the phone happily gets their dialtone.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 lines
Several chan_zap options were not working on reload because they were arbitrarily
disallowed when reloading some/most PRI options (such as signalling) was disallowed.
Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload.
This corrects that behavior.
Issue 9186, patch by tzafrir.
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variable declarations in the middle of a block.
Fix the few instances of the above spotted out by the compiler.
All of this has been already done or is not applicable in trunk,
so the merge of this change will be blocked.
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specifies 'group=42' it will actually work instead of being silently ignored
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | 5 lines
Fix a problem where an established call would not be properly disconnected
when a PRI disconnect is received depending on which cause code was received.
(issue #9588, original patch by softins, updated patch from jtexter3, and some
additional feedback from mhardeman)
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reported by premeau)
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Issues 9808 and 9809.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines
Revert revision 62417 as someone reported problems with it to Mark. This was
related to issue #9588.
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(Thanks to mnicholson for pointing it out)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588, reported and patched by softins)
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functional. (issue #9626, reported and fixed by sergee)
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option_debug (DEA). Thanks!
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587, reported and patched by xrg)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line
removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()
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refinement, but this won't have as many folks bothered.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines
There should only be one code path for doing DTMF conditionals on channels. This fixes it.
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incorrectly destroy the channel. (issue #9242, andrew)
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(issue #7256, tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines
When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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while in use will not result in crashes. (issue #8897 reported by junky)
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The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file.
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offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)
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r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines
Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer)
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