Commit graph

2725 commits

Author SHA1 Message Date
Mark Michelson
dea49116ee Merged revisions 92323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec 2007) | 10 lines

Fixing autofill to be more accurate. Specifically, if calls ahead of the current
caller were ringing members (but not yet bridged) there could be available members
and waiting callers who would not get matched up. The member availability checker
was correctly determining the number of available members in this scenario, but
the queue itself did not parallelly reflect this status on the pending calls. This
commit corrects the issue.

(closes issue #11459, reported by equissoftware, patched by me)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-11 17:44:42 +00:00
Mark Michelson
8bf68432a0 Merged revisions 92202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec 2007) | 7 lines

If there are no members in a queue, then the loop where the datastore for detecting
duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means
that when we try to free it, there's a crash. This stops that crash from occurring.

(closes issue #11499, reported by slavon, patched by eliel)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-10 16:30:46 +00:00
Olle Johansson
e2a8a6f46a Add a few extra headers in the voicemail users listing in
manager 1.1. Update documentation too.

(closes issue #11495)
Reported by: caio1982
Patches: 
      extra_vm_manager_info1.diff uploaded by caio1982 (license 22)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-10 13:29:57 +00:00
Luigi Rizzo
5490889153 Put into Makefile.moddir_rules the common instructions used to
generate loadable and embedded module lists.

Individual Makefiles now are a lot simpler, possibly as simple as this:

    -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
    MODULE_PREFIX=cdr_
    all: _all
    include $(ASTTOPDIR)/Makefile.moddir_rules

and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.

The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).

With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-10 03:50:38 +00:00
Luigi Rizzo
d652be0930 normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference
the top level directory.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-09 21:29:37 +00:00
Russell Bryant
3a4d1c852b Merged revisions 91783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines

* Add channel locking around datastore operations that expect the channel
  to be locked.
* Document why we don't record Local channels in the dialed interfaces list.
* Remove the dialed variable as it isn't needed.
* Restructure some code for clarity and coding guidelines stuff

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2007-12-07 16:40:41 +00:00
Russell Bryant
bfd58d8c2a Merged revisions 91780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | 7 lines

* Add channel locking around datastore operations that expect the channel
  to be locked.
* Document why we don't record Local channels in the dialed interfaces list.
* Handle memory allocation failure.
* Remove the dialed variable, as it wasn't actually needed.
* Tweak some formatting to conform to coding guidelines.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-07 16:28:36 +00:00
Russell Bryant
547083e21a Merged revisions 91693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | 2 lines

Don't unlock the dialed_interfaces list until we're done messing with the iterator.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-07 02:52:38 +00:00
Russell Bryant
c72fa81580 Merged revisions 91677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | 4 lines

Allow dialing local channels from Queue() and Dial() again.  There was a slight
flaw in the code to prevent call forwards from looping that caused this problem.
(related to issue #11486)

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2007-12-07 02:43:21 +00:00
Russell Bryant
135f315382 Merged revisions 91675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | 7 lines

Fix in an issue in the call forwarding handling code that was causing crashes
on every call into a queue.  I'm not entirely sure about the logic in this part
of the code, so I want to look at it some more tomorrow.  However, this makes
it safe and keeps it from crashing.

(closes issue #11486, reported by adamg, patched by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-07 02:21:07 +00:00
Mark Michelson
e797cd04dc Handle allocation failure of the heard and deleted arrays of the vm_state.
(closes issue #11408, reported and patched by jaroth)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06 21:03:15 +00:00
Olle Johansson
807d5e1ef7 - Dial event
- Event Dial has new headers, to comply with other events
        - Source        -> Channel              Channel name (caller)
        - SrcUniqueID   -> UniqueID             Uniqueid
        (new)           -> Dialstring           Dialstring in app data


(moremanager)


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2007-12-06 15:04:34 +00:00
Olle Johansson
6765d5f758 Adding small missing but important comma...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06 14:55:31 +00:00
Olle Johansson
478df6c69f A big oops...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06 14:40:44 +00:00
Olle Johansson
a545aaae53 The MeetmeJoin now has caller ID name and Caller ID number fields (like MeetMeLeave)
(Moremanager)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06 14:36:54 +00:00
Mark Michelson
fe83f51186 Merged revisions 91292 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec 2007) | 3 lines

Reverting extra stuff I didn't mean to commit


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05 22:57:57 +00:00
Mark Michelson
b32e39cbda Merged revisions 91273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines

The 'G' option for Dial() did not properly handle the case where only a label was
provided. This was due to the fact that the answering channel did not have an extension
set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto
on the answering channel since it is a wasteful call. The answering channel and the calling
channel are both directed to the same extension and context, just different priorities, so
we can just copy the values from the calling channel to the answering channel and increment
the answering channel's priority.

(closes issue #11382, reported by jon, patch by me with correction by jon)


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2007-12-05 22:55:49 +00:00
Russell Bryant
f82c42a22e Resolve compiler warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05 17:44:59 +00:00
Tilghman Lesher
d226c1d637 Added multiple name listing. (Closes issue #10413)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05 16:25:52 +00:00
Mark Michelson
0742acef39 Kevin suggested doing the reverse of my last commit, since imap_retrieve_file
does not modify the contents of the "mailbox" string. In other words, I'm changing
the imap_retrieve_file function to take a const char* as the third argument so that I
don't need to cast const char*'s as char*'s to suppress compiler warnings.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 18:29:35 +00:00
Mark Michelson
50ee083210 Suppress a compiler warning due to discarding a "const" qualifier
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 18:14:08 +00:00
Mark Michelson
5d1fb935ba Wrong locking style got merged from 1.4 to trunk. My mistake.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:51:59 +00:00
Jason Parker
814a7f66c0 Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this hadn't been merged yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:35:40 +00:00
Mark Michelson
c52d8a1cd5 Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines

A big one...

This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.

This change also introduces some side effects to the code which I shall enumerate here:

1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
   which handles the call forward case after the channel has been requested but before it has
   been called. This was removed because call-forwarding still works fine without it, it makes the
   code less error-prone should it need changing, and it made this set of changes much less painful
   to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
   which is attached to the channel may be created and attached in either app_dial or app_queue, so they
   need a common place to find the datastore info. This approach was taken in case similar datastores are
   needed in the future, there will be a common place to add them.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:08:36 +00:00
Olle Johansson
4d2368f202 (closes issue #11431)
Reported by: Laureano
Patches: 
      app_queue.c.patch uploaded by Laureano (license 265)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 15:16:03 +00:00
Jason Parker
d3dd515072 Merged revisions 90696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

(Closes issue #11383)
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r90696 | qwell | 2007-12-03 16:06:36 -0600 (Mon, 03 Dec 2007) | 4 lines

Make sure we always close the conference fd if we have an open one.

Issue 11383, reported by markmhy, patch by eliel.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03 22:07:57 +00:00
Mark Michelson
9c2b82c726 Replacing some calls to free() with ast_free().
(closes issue #11448, reported and patched by jaroth)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03 21:24:56 +00:00
Joshua Colp
4201a5af8b Remove the file descriptors from the main poll channel when the channel is hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end.
(closes issue #11441)
Reported by: blitzrage


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03 14:14:43 +00:00
Russell Bryant
0f5117be2e Merged revisions 90470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | 6 lines

The other day when I went through making changes as a result of the ao2_link()
change, I added some code to set pointers to NULL after they were unreferenced.
This pointed out that in this place, the object was unreferenced before the
code was done using it.  So, move the unref down a little bit.
(crash reported by jmls on IRC)

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2007-12-02 18:20:13 +00:00
Mark Michelson
6b08c442c7 Adding support for the "automixmonitor" dial and queue options.
This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.

This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.

Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to 
be committed.

(closes issue #10185, reported and patched by xmarksthespot)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-30 21:19:57 +00:00
Russell Bryant
fac7480820 Merged revisions 90348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines

Change the behavior of ao2_link().  Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference
that is now held by the container.

This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container.  It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.

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2007-11-30 19:34:47 +00:00
Mark Michelson
4ed5336a45 Merged revisions 90163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov 2007) | 6 lines

This patch handles the case where a queue member with a negative penalty is added
via the manager. If a negative value is submitted for a member penalty, we set it to 0.

(closes issue #11411, reported and patched by Laureano)


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2007-11-29 19:39:31 +00:00
Joshua Colp
48da910225 Merged revisions 90101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 lines

Fix a few memory leaks.
(closes issue #11405)
Reported by: eliel
Patches:
      load_realtime.patch uploaded by eliel (license 64)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28 23:03:09 +00:00
Russell Bryant
972cacad4a Merge some little changes from team/russell/chan_refcount to help reduce
the diff to trunk.

This just removes some checks on the return value of alloca(), as behavior
is undefined if it runs out of stack space, and we don't check it anywhere else.


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2007-11-28 00:49:55 +00:00
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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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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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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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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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?


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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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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