the beginning of the file. Also, add a channel variable that indicates
the location in the file where the Playback was stopped.
(closes issue #7655, patch from sharkey)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r68198 | mmichelson | 2007-06-07 14:47:42 -0500 (Thu, 07 Jun 2007) | 5 lines
Submitting a fix for Issue 8016. Added a check to make sure that greetings get stored properly.
(Issue 8016, reported by edhorton, patched by alamantia with modification by me. Thanks to Jason Parker
for the advice on this).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r67804 | mmichelson | 2007-06-06 14:26:55 -0500 (Wed, 06 Jun 2007) | 10 lines
Fix for Issue 9810. There was a segfault under a specific set of circumstances:
1. VoiceMailMain was configured in the dialplan with an extension as its argument
2. A message was left for this mailbox
3. Tried to call VoiceMailMain but hung up before entering password.
This was fixed by checking that a pointer was non-null prior to trying to dereference it.
(Issue 9810, reported by xmarksthespot, patched by Corydon76 with modifications by me).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
was enabled. Though no bug was reported to the bugtracker, there was mention of this made as a note on
bug 9810 by edhorton.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) | 5 lines
Fix some crashes related to the use of the "meetme" CLI command. The code for
this command was not locking the conference list at all.
(issue #9351, reported by and patch submitted by Junk-Y, committed patch
is different and by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r67424 | mmichelson | 2007-06-05 13:32:50 -0500 (Tue, 05 Jun 2007) | 5 lines
Fix for bug number 9786, wherein voicemails saved to IMAP storage using extensions other than gsm were
unable to be played over the phone. (Issue 9786, reporter: xmarksthespot, Patched by xmarksthe spot with revisions by me,
reviewed by Russell Bryant).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67486 65c4cc65-6c06-0410-ace0-fbb531ad65f3
places that cared about device states were app_queue and the hint code in pbx.c.
The changes include converting it to use the Asterisk event system, as well as
other efficiency improvements.
* app_queue: This module used to register a callback into devicestate.c to
monitor device state changes. Now, it is just a subscriber to Asterisk
events with the type, device state.
* pbx.c hints: Previously, the device state processing thread in devicestate.c
would call ast_hint_state_changed() each time the state of a device changed.
Then, that code would go looking for all the hints that monitor that device,
and call their callbacks. All of this blocked the device state processing
thread. Now, the hint code is a subscriber of Asterisk events with the
type, device state. Furthermore, when this code receives a device state
change event, it queues it up to be processed by another thread so that it
doesn't block one of the event processing threads.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r66897 | mmichelson | 2007-06-01 16:09:30 -0500 (Fri, 01 Jun 2007) | 3 lines
Submitting a fix for voicemail with IMAP storage. Attachments with format specified as gsm were duplicated (i.e. two attachments) were left.
Thank you very much to xmarksthespot for submitting the patch that fixed this. (Issues 9787 and 8873, Reported by xmarksthespot and jerjer, patched by xmarksthespot)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Don't free structures before calling load_config(), because load_config()
already does it
- Use the existing functions for freeing the minivm structures instead of
replicating the code
(issue #9846, patch from eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is no reason to keep a thread attribute structure on the conference structure.
(Pointed out by Tony Mountifield on the asterisk-dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
saw this, I couldn't help myself from changing it. Previously, for *every*
device state change, app_queue would spawn a thread to handle it. Now, the
device state callback just puts the state change in a queue and it gets
handled by a single state change processing thread.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) | 5 lines
Fix a small bug I noticed while working on something else. app_queue did not
unregister its device state monitoring callback in unload_module(). So, this
would make Asterisk crash on the first device state change after you
unload the module.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
