https://origsvn.digium.com/svn/asterisk/trunk
........
r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug 2008) | 18 lines
Merging the imap_consistency_trunk branch to
trunk.
For an explanation of what "imap_consistency" is,
please see svn revision 134223 to the 1.4 branch.
Coincidentally, this also fixes a recent bug report
regarding the inability to save messages to the new
folder when using IMAP storage since they will would
be flagged as "seen" and not be recognized as new
messages.
(closes issue #13234)
Reported by: jaroth
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines
Merged revisions 135799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r135067 | mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 lines
IMAP storage functioned under the assumption that folders
such as "Work" and "Family" would be subfolders of the
INBOX. This is an invalid assumption to make, but it could
be desirable to set up folders in this manner, so a new
option for voicemail.conf, "imapparentfolder" has been
added to allow for this.
(closes issue #13142)
Reported by: jaroth
Patches:
parentfolder.patch uploaded by jaroth (license 50)
........
r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug 2008) | 3 lines
IMAP-specific items must go in IMAP_STORAGE defines...
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul 2008) | 24 lines
Merged revisions 134758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines
Add more timeout checks into app_queue, specifically
targeting areas where an unknown and potentially
long time has just elapsed. Also added a check
to try_calling() to return early if the timeout
has elapsed instead of potentially setting a negative
timeout for the call (thus making it have *no* timeout
at all).
(closes issue #13186)
Reported by: miquel_cabrespina
Patches:
13186.diff uploaded by putnopvut (license 60)
Tested by: miquel_cabrespina
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines
Merged revisions 133169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines
Remove libresample from the Asterisk source tree. It is now available in its
own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r131824 | mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 lines
Document that the duration of dtmf may be passed to
the SendDTMF application. Also correct the default
pause between digits.
(closes issue #13102)
Reported by: eliel
Patches:
app_senddtmf.c.patch uploaded by eliel (license 64)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul 2008) | 22 lines
Merged revisions 131369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
(closes issue #13084)
Reported by: elbriga
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul 2008) | 21 lines
Merged revisions 131299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
(closes issue #13047)
Reported by: festr
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
Merging this rev from trunk to 1.6.0 was not
simple. Why? Because we've enhanced trunk to
do a [fast] merge-and-delete operation which
also solved problems with contexts having
entries from different registrars.
Fast as in the amount of time the contexts
are locked down. That *is* fast, but traversing
the entire dialplan looking for priorities to
delete takes more time overall.
This particular fix involved pulling in those
enhancements from trunk, along with all the
various fixes and refinements made along the
way.
Merging all this from trunk into 1.6 involved:
a. mergetrunk6 in the stuff from 130145;
b. revert all but the prop changes
c. catalog all revisions to pbx.c since 1.6.0 was forked
(at rev 105596).
d. catalog all revisions to pbx.c in trunk since 1.6.0
was forked, making special note of all revs that
were not merged into 1.6.0.
e. study each rev in trunk not applied to 1.6.0, and
determine if it was involved in the merge_and_delete
enhancements in trunk. 25 commits were done in 1.6.0,
all but one (106306) was a merge from trunk.
Trunk had 22 additional changes, of which 7 were
involved in the merge_and_delete enhancements:
106757
108894
109169
116461
123358
130145
130297
f. Go to trunk and collect patches, one by one,
of the changes made by each rev across the
entire source tree, using svn diff -c <num> > pfile
g. Apply each patch in order to 1.6.0, and
resolve all failures and compilation problems
before proceding to the next patch.
h. test the stuff.
i. profit!
........
r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines
(closes issue #13041)
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines
Merged revisions 130792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r129684 | bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines
Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the
state_interface isn't stored in the ast_db.
(closes issue #13043)
Reported by: jvandal
Patches:
app_queue.patch uploaded by jvandal (license 413)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@129685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r129006 | russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines
Update app_fax for better compatibility with spandsp 0.0.5. Add a call to
t38_terminal_release, and make sure that the phase E handler gets called
with proper status.
(closes issue #13020)
Reported by: dimas
Patches:
v1-appfax.patch uploaded by dimas (license 88)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@129007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul 2008) | 7 lines
If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain,
app_voicemail would attempt to play a file called vm-foo instead of playing
vm-INBOX to play the "new" sound file. This commit fixes that issue.
This may fix one of the problems reported in issue #12987
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@128732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r125649 | mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15 lines
The monitor-join option for queues was deprecated in favor of using
MixMonitor to mix audio. However, it was pointed out to me that because
of this, the command set for the MONITOR_EXEC variable is ignored as well.
This means that people can't do their own custom mixing commands at the end
of recordings in order to make, for instance, stereo recordings of calls.
With this patch, app_queue will set the "joinfiles" variable for the channel's
monitor if MONITOR_EXEC is not zero-length. This means that for normal audio
mixing, MixMonitor is still the preferred choice, but we allow custom
mixing to be done with the two Monitor streams if desired.
(closes issue #12923)
Reported by: snyfer
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun 2008) | 19 lines
Merged revisions 125585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines
Add the interface of a queue member to the output of the "queue show" command
so that it can easily be associated with a queue member's name. This helps
so that the appropriate queue member can be removed or paused since the
interface is required, not the member's name.
(closes issue #12783)
Reported by: davevg
Patches:
app_queue.diff uploaded by davevg (license 209) with small mod from me
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
