Commit graph

2691 commits

Author SHA1 Message Date
Russell Bryant
63bca744a2 Merged revisions 89844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines

Instead of depending on the return value of ast_true(), explicitly set the
eventwhencalled variable to 1.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27 23:21:38 +00:00
Mark Michelson
ba7f5fec38 Merged revisions 89837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines

Two changes with regards to the 'eventwhencalled' option of queues.conf

1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 
   'vars' or 'yes' did exactly the same thing. Thus the sign change of the
   ast_true call.

2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting
   in bizarre output for the channel variables. This patch remedies this.

(related to issue #11385, however I'm not sure if this will actually be enough to close it)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27 23:11:12 +00:00
Steve Murphy
4d8932a6dc Merged revisions 89622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27 06:47:08 +00:00
Mark Michelson
5f3a28e588 Merged revisions 89618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines

After issuing a "say load new", if a caller hangs up during the middle of playback of a number,
app_playback will continue to try to play the remaining files. With this change, no more files will
be played back upon hangup.

(closes issue #11345, reported and patched by IgorG)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 23:11:29 +00:00
Olle Johansson
130a2051fa - Mark "concise" as deprecated
- 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
2007-11-26 19:24:23 +00:00
Joshua Colp
0619fb1248 Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 18:11:31 +00:00
Joshua Colp
9905034266 Merged revisions 89587 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 17:23:28 +00:00
Kevin P. Fleming
721b3c8a0e Merged revisions 89586 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 17:21:37 +00:00
Mark Michelson
0b120dac62 Merged revisions 89580 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 15:50:37 +00:00
Joshua Colp
5303abd58d Merged revisions 89571 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 14:42:57 +00:00
Joshua Colp
2c223a4512 Don't crash if the 'o' option of ControlPlayback is used without any value.
(closes issue #11375)
Reported by: johan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 14:31:32 +00:00
Tilghman Lesher
b0d8378910 Merged revisions 89540 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-24 06:24:46 +00:00
Luigi Rizzo
cda3df64d8 more header removal
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22 04:37:08 +00:00
Luigi Rizzo
51391e6b09 shuffle a little bit the content of header files to reduce dependencies.
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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22 03:50:04 +00:00
Luigi Rizzo
200f9c633b remove some useless includes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22 02:30:58 +00:00
Luigi Rizzo
ea2c54859d more removal of redundant headers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22 02:07:33 +00:00
Luigi Rizzo
b1fe2d85d3 remove redundant headers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22 01:39:06 +00:00
Luigi Rizzo
3fc2646808 remove a number of #include <fcntl.h> which are either
useless or done elsewhere



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22 01:03:02 +00:00
Steve Murphy
86476c607f closes issue #11285, where an unload of a module that creates a dialplan 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:54:12 +00:00
Luigi Rizzo
7e8835e0d7 remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:24:55 +00:00
Mark Michelson
ffa8035212 Merged revisions 89495 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 19:28:43 +00:00
Mark Michelson
9c275b0e51 Merged revisions 89493 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 19:27:22 +00:00
Tilghman Lesher
1c295be7a0 Change Read to set READSTATUS as an indication of the result
Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 18:38:18 +00:00
Russell Bryant
6335b4b30d Merge changes from team/russell/sla_trunk_moh ...
* 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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 00:21:38 +00:00
Tilghman Lesher
cbfc6dcbea Make trunk build again
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20 23:29:33 +00:00
Luigi Rizzo
a23c055c3d move asterisk/paths.h outside asterisk.h and into those files
who really need it.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20 23:16:15 +00:00
Luigi Rizzo
6938f4b2b0 Fix building of modules under cygwin.
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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20 16:12:10 +00:00
Luigi Rizzo
ffd86fc964 more errno.h removal
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 21:12:08 +00:00
Luigi Rizzo
0595b5e2aa include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere.
Remove some of the instances.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 18:52:04 +00:00
Joshua Colp
23cfef1cc9 Change warning messages (which are really debug messages) into debug messages.
(closes issue #11288)
Reported by: IgorG
Patches:
      saydebug-89394-1-trunk.patch uploaded by IgorG (license 20)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 14:03:30 +00:00
Luigi Rizzo
24c2e47e0d another cygwin compatibility fix.
This one must be handled in a better way in configure, also for other
architectures



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 11:08:58 +00:00
Luigi Rizzo
d82a631f9c more removal of duplicate #include lines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 00:02:33 +00:00
Luigi Rizzo
5490960453 remove a bunch of duplicate includes
Reproduce with

grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr 



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 23:54:45 +00:00
Luigi Rizzo
fdb7f7ba3d Start untangling header inclusion in a way that does not affect
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.).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Mark Michelson
68a805b70b Adding confirmation playback when forwarding voicemail messages. This will 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)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 17:11:43 +00:00
Mark Michelson
2f7440932c Merged revisions 89323 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 15:44:27 +00:00
Tilghman Lesher
8044dce3c0 Fix trunk breakage due to chan->lock being renamed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15 18:15:48 +00:00
Russell Bryant
1bf63d1c5b Merged revisions 89296 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15 17:27:27 +00:00
Luigi Rizzo
09d9cce1d8 access channel locks through ast_channel_lock/unlock/trylock and not
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).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15 16:20:47 +00:00
Luigi Rizzo
7f8ecd2cd3 make the 'name' and 'value' fields in ast_variable const char *
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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 13:18:40 +00:00
Russell Bryant
0df5e50e97 Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor happy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 01:40:47 +00:00
Russell Bryant
31512895ae Instead of reserving 800 bytes for periodic announcements, use an array of
ast_str pointers and only alloate space for the strings as needed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 01:35:28 +00:00
Russell Bryant
76696ac65f Convert most of the strings in the call_queue struct to use stringfields.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 01:15:26 +00:00
Mark Michelson
e4bb565530 Merged revisions 89241 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13 16:03:10 +00:00
Mark Michelson
13c49f6cce There is the potential to copy uninitialized memory into the mixmonitor->post_process
string. This fix prevents that.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13 01:19:53 +00:00
Mark Michelson
1d4c579422 Merged revisions 89205 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13 00:57:34 +00:00
Steve Murphy
98429d37b2 Based on a note in asterisk-dev by Brian Capouch, I determined I too 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12 18:44:36 +00:00
Mark Michelson
35d7946cda app_voicemail failed to build when compiling with IMAP_STORAGE
Now it does not.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08 22:33:59 +00:00
Tilghman Lesher
3a70afbc3e Add the FILE() dialplan function and deprecate ReadFile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08 17:32:15 +00:00
Luigi Rizzo
01a1cfd262 use %f instead of %lf (the 'l' is ignored anyways).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08 09:21:02 +00:00