not be loaded from realtime queues. This commit fixes that.
Thanks to jmls for pointing this problem out to me on IRC.
This also contains some changes to S_OR where it should be used. Thanks to Qwell for pointing
these out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is done in a backward compat way.
If the "default" key for ffwd/rew is used for another option (such as stop), the "default" is removed.
(closes issue #11754)
Reported by: johan
Patches:
app_controlplayback.c.option3.patch uploaded by johan (license 334)
Tested by: johan, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan 2008) | 5 lines
Fix a logic error with regards to autofill. Prior to this change, it was possible
for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting
to call a member. This change fixes this.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | 8 lines
ChanSpy issues a beep when it starts at the beginning of a list of channels to
potentially spy on. However, if there were no matching channels, it would beep
at you over and over, which is pretty annoying. Now, it will only beep once in
the case that there are no channels to spy on, but it will still beep again once
it reaches the beginning of the channel list again.
(closes issue #11738, patched by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) | 8 lines
When we reset the password via an external command, we should also reset the
password stored in the in-memory list, too (otherwise it doesn't really take
effect).
(closes issue #11809)
Reported by: davetroy
Patches:
fix_externpass.diff uploaded by davetroy (license 384)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote
dimas from the original bug description:
"app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences.
1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be.
2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa).
3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message.
4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list.
5. Alot of duplicated code as already mentioned."
This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen
in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is
well worth it.
Huge thanks to dimas for this wonderful submission.
(closes issue #11744)
Reported by: dimas
Patches:
dir3.patch uploaded by dimas (license 88)
Tested by: putnopvut, dimas
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines
Adding explicit defaults for missing options to init_queue. This is necessary because
if a user either removes or comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the value from prior to the
reload.
Thanks to John Bigelow for pointing this error out to me.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
option tells JACK not to start jackd automatically if it is not already
running. Otherwise, the default is that jackd will get started for you if
it isn't running already.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/). I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines
Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
