https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov 2007) | 13 lines
Removing some seemingly pointless code. This sets a channel variable for every priority
executed in the dialplan if you have debug set to anything non-zero. This seems pointless
due to the fact that these channel variables are not referenced anywhere else in the code and
their names are esoteric enough that they would not be practical to reference in the dialplan. Plus
the fact that this behavior isn't documented anywhere means that the change is not likely to cause
any disruption. If anything, this may actually cause a slight performance increase if running with
debug on.
The motivating influence for this code change is the eventwhencalled option for queues. If set to
vars, all channel variables will be output to the manager. These unnecessary channel variables make
the output a lot more difficult to deal with.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines
- update documentation for some of the goto functions to note that they
handle locking the channel as needed
- update ast_explicit_goto() to lock the channel as needed
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines
Merge changes from team/russell/autoservice_1.4
This set of changes fixes an issue that was reported to me on IRC yesterday.
The user, d1mas, was using chan_zap for incoming calls and was having DTMF
recognition issues in some situations. Specifically, he noticed that the
problem occurred when using DISA or WaitExten. He also noticed that when
using Read, the problem did not occur. His system also used DUNDi for
dialplan lookups.
So, he theorized that if the DUNDi lookups blocked for some period of time,
that audio from the zap channel could get lost. If the audio got lost, then
it wouldn't be run through the DTMF detector, and digits could get lost.
He was correct, and the following set of changes fixes the problem. However,
the changes go a little bit further than what was necessary to fix this exact
problem.
1) I updated pbx_extension_helper() to autoservice the associated channel to
handle cases where extension lookups may take a long time. This would
normally be a dialplan switch that does some lookup over the network, such
as the DUNDi or IAX2 switches.
This ensures that even while a DUNDi lookup is blocking, the channel will be
continuously serviced.
2) I made a change to the autoservice code. This is actually something that
has bothered me for a long time. When a channel is in autoservice, _all_
frames get thrown away. However, some frames really shouldn't be thrown
away. The most notable examples are signalling (CONTROL) frames, and DTMF.
So, this patch queues up important frames while a channel is in autoservice.
When autoservice is stopped on the channel, the queued up frames get stuck
back on the channel so that they can get processed instead of thrown away.
3) I made another change to the autoservice code to handle the case where
autoservice is started on channels recursively.
Previously, you could call ast_autoservice_start() multiple times on a
channel, and it would stop the first time ast_autoservice_stop() gets
called. Now, it will ensure that autoservice doesn't actually stop until
the final call to ast_autoservice_stop().
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines
on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...
Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines
Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines
If channel allocation fails because the alert pipe could not be created also free the scheduler context.
(closes issue #11355)
Reported by: eliel
Patches:
main.channel.c.patch uploaded by eliel (license 64)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines
Currently, zero-length voicemail messages cause a hangup in VoicemailMain.
This change fixes the problem, with a multi-faceted approach. First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007) | 10 lines
Up until this point, the XML output of the manager has been technically
invalid, due to the repetition of certain parameters in a single event.
This caused various issues for XML parsers, some of which refused to parse
at all, given the invalidity of the rendered XML. So this commit fixes
the XML output, ensuring that each entity parameter has a unique name, thus
ensuring valid XML.
Reported by: msetim
Patch by: tilghman
(Closes issue #10220)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
only do the calculations if fax detection is enabled on the dsp.
(closes issue #11331)
Reported by: dimas
Patches:
dsp.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
would return LONG_MIN (1 in 9 quintillion if using 64-bit longs). Since there
is no positive equivalent of LONG_MIN, the result of labs() in this case is
unpredictable. This fixes that situation.
(closes issue #11336, reported and patched by sperreault)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unfortunately, since trunk uses read/write locks for the context lock, it means that I have
actually *introduced* a deadlock condition since they are not recursive. Removing this change
for now and will look into introducing a different one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
