- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115588 f38db490-d61c-443f-a65b-d21fe96a405b
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114813 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110629 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb 2008) | 10 lines
There was an invalid assumption when calculating the duration of a file that the filestream in question
was created properly. Unfortunately this led to a segfault in the situation where an unknown format was
specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream
was written correctly or else assume a zero duration.
(closes issue #12021)
Reported by: jakep
Tested by: putnopvut
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103787 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines
on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89721 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89541 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
- 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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007) | 3 lines
Making change to group splitting, as discussed on the -dev list. The main
effect of this will be to permit Set(GROUP([cat])=), i.e. unsetting a group.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83654 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81233 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@68502 f38db490-d61c-443f-a65b-d21fe96a405b
