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(closes issue #11335)
Reported by: rebuild
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is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
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on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
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other endpoint, ensure that any DSP active on the original channel is moved to the new one
(closes issue #11917)
Reported by: mavetju
Tested by: mavetju
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still
set to make sure that when we come back out of alarm, it gets reported in the log
and manager interface (after discussion with tzafrir on the -dev list)
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does not know what to do with these alarms. Only Asterisk 1.6 cares about it.
So, if we get an unknown alarm in chan_zap, don't generate confusing log messages
about it.
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This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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calls from a section of code where we know there was no event to handle or get.
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specific event and tried to report
which event was received. What actually was happening was that it was reporting the number of bytes returned
from a call to read().
Thanks to Jared Smith for bringing the issue up on IRC
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setup message contains the sending_complete information element.
(closes issue #11785)
Reported by: klaus3000
Patches:
sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by klaus3000 (license 65)
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is off.
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Issue #12019
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could lead to a looped list, meaning it will traverse forever.
(closes issue #11818)
Reported by: michael-fig
Patches:
11818.patch uploaded by putnopvut (license 60)
Tested by: michael-fig
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(closes issue #11590)
Reported by: tbsky
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(closes issue #11675)
Reported by: flefoll
Patches:
chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244)
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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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