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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines
Close the audio file before sending it to the post processing application.
(closes issue #11357)
Reported by: reformed
Patches:
mixmonitor.patch uploaded by reformed (license 330)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines
when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines
Revert vmu->email back to an empty string if it was empty when imap_store_file
was called. This prevents sending a duplicate e-mail.
(closes issue #11204, reported by spditner, patched by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 lines
When unloading app_meetme destroy any auto created contexts created by SLA.
(closes issue #11367)
Reported by: eliel
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(closes issue #11375)
Reported by: johan
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
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useless or done elsewhere
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context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov 2007) | 3 lines
Fix a small error I made in my previous commit
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov 2007) | 5 lines
Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this
message would always report that there were 0 members available, even though that may not be true.
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Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)
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* Added the ability to specify the music on hold class used to play into the
conference when there is only one member and the M option is used.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
(patched by me, and tested internally)
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who really need it.
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After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
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were included almost everywhere.
Remove some of the instances.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11288)
Reported by: IgorG
Patches:
saydebug-89394-1-trunk.patch uploaded by IgorG (license 20)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89410 f38db490-d61c-443f-a65b-d21fe96a405b
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This one must be handled in a better way in configure, also for other
architectures
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Reproduce with
grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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attempt
to play the name(s) of the person(s) to whom you are forwarding the message prior to
prompting for prepending. If no name is found, the extension is read back verbatim.
(closes issue #9046, reported and patched by jaroth)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov 2007) | 5 lines
Make realtime queues accessible from the QUEUE_MEMBER_COUNT function.
(closes issue #11271, reported and patched by atis, with small modifications from me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | 8 lines
Update the SLAStation application to account for the case where the SLA thread
has a call out to the station, but the user has pressed a line button to answer
the call instead of picking up the handset. If they do, the phone sends out a
new INVITE. So, the SLAStation app must check to see if it is picking up a
ringing trunk, and ensure that the other stations stop ringing.
(reported internally, patched by me, tested by mogorman)
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through ast_mutex primitives.
To detect all occurrences, I have renamed the lock field in struct ast_channel
so it is clear that it shouldn't be used directly.
There are some uses in res/res_features.c (see details of the diff)
that are error prone as they try and lock two channels without
caring about the order (or without explaining why it is safe).
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This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
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happy
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ast_str pointers and only alloate space for the strings as needed.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines
Reverting commit made in revision 89205 since it is unnecessary.
Thanks to Kevin for pointing this out
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mixmonitor->post_process
string. This fix prevents that.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines
Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options
and args.post_process strings are uninitialized and could contain garbage. This change
handles this situation properly by only using arguments that we have parsed.
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agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string.
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Now it does not.
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) | 7 lines
The member refcount must be incremented, to avoid using it after deallocation.
A huge thanks go to lvl- for patiently providing the necessary valgrind output
that was necessary to finding this problem of memory corruption.
Reported by: lvl-
Patch by: tilghman
Closes issue #11174
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INBOX...
ever. It may not exist.
(closes issue #11151, reported by selsky, patched by me)
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Reported by: eliel
Patch by: eliel
Closes issue #11177
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This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
(closes issue #11078)
Reported by: jthomas
Patches:
meetme-concise.patch uploaded by jthomas (license 293)
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(closes issue #10942, reported and patched by julianjm, documentation changes by me)
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