https://origsvn.digium.com/svn/asterisk/branches/10
................
r330051 | rmudgett | 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) | 29 lines
Merged revisions 330050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines
Merged revisions 330033 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines
Datacalls with B410P fail.
Incoming and outgoing call legs of a data call are using different
formats: a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation is bad
for data calls.
* Make incoming call that does not explicitly specify u-law or a-law use
the DAHDI channel's default law. The outgoing call always uses the
default law from the DAHDI channel.
(closes issue ABE-2800)
Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett
..........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
........
r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu, 28 Jul 2011) | 4 lines
The default conf-usermenu says that '8' can be used to leave the conference, so
put that in the sample user menu. '5' is supposed to extend the conference, but
there doesn't appear to be a concept of that in the menu actions.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r329538 | jrose | 2011-07-26 09:19:34 -0500 (Tue, 26 Jul 2011) | 11 lines
Merged revisions 329529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | 5 lines
Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same
prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more
appropriate anyway.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines
Merged revisions 329527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines
Fixes some voicemail forwarding behavior based around prepend mode.
Formerly, prepend forwarding would have the user record a message with no useful prompt
and an expectation for the user to push a button on the phone when finished recording.
If a length of silence was detected instead, the recording would be canceled and the user
would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording
would also bug out in the sense that they would write over the original message and get
sent to the recipient regardless of whether they timed out or were accepted. This patch
fixes this issue and adds a prompt which will be played after a timeout informing the
user that they needed to press a button. Currently, the sound files that we have are
somewhat inadquate for this, so after the call we simply have Allison say "Please try
again. Then press pound." which actually relies on two separate sound files. Just one
would be more appropriate.
reporter: Vlad Povorozniuc
Review: https://reviewboard.asterisk.org/r/1327/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r329331 | rmudgett | 2011-07-22 15:43:07 -0500 (Fri, 22 Jul 2011) | 55 lines
Merged revisions 329299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines
Deadlocks dealing with dialplan hints during reload.
There are two remaining different deadlocks reported dealing with dialplan
hints.
The deadlock in ASTERISK-17666 is caused by invalid locking order in
ast_remove_hint(). The hints container must be locked before the hint
object.
The deadlock in ASTERISK-17760 is caused by a catch-22 situation in
handle_statechange(). The deadlock is caused by not having the conlock
before calling the watcher callbacks. Unfortunately, having that lock
causes a different deadlock as reported in ASTERISK-16961.
* Fixed ast_remove_hint() locking order.
* Made handle_statechange() no longer call the watcher callbacks holding
any locks that matter.
* Made hint ao2 destructor do the watcher callbacks for extension
deactivation to guarantee that they get called.
* Fixed hint reference leak in ast_add_hint() if the callback container
constructor failed.
* Fixed hint reference leak in complete_core_show_hint() for every hint it
found for CLI tab completion.
* Adjusted locking in ast_merge_contexts_and_delete() for safety.
* Added context_merge_lock to prevent ast_merge_contexts_and_delete() and
handle_statechange() from interfering with each other.
* Fixed ast_change_hint() not taking into account that the extension is
used for the hash key.
(closes issue ASTERISK-17666)
Reported by: irroot
Tested by: irroot
JIRA SWP-3318
(closes issue ASTERISK-17760)
Reported by: Byron Clark
Tested by: irroot
JIRA SWP-3393
Review: https://reviewboard.asterisk.org/r/1313/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines
Merged revisions 329199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines
Update PickupChan documentation.
The PickupChan uses the ampersand as the argument separator.
Was documented as:
PickupChan(channel[,channel2[,...][,options]])
Fixed documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144)
Reported by: Erik Smith
Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
Tested by: Erik Smith
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/2.0
................
r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
Merged revisions 328935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
Inband DTMF regression
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/2.0
................
r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines
Merged revisions 328878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines
Revert partial attempt at handling pathnames with spaces.
Revision 299794 attempted to improve the build system to be able to handle
pathnames (primarily DESTDIR) with spaces in them, since this is common on
some platforms (including Mac OSX). Unfortunately, the changes were incomplete
and did not actually provide the desired behavior, and as a side effect the
functionality that ensured that stale headers in the Asterisk 'include' directory
were removed got broken. In addition, the check for stale (and possibly
incompatible) modules in the Asterisk 'modules' directory also got broken, and
would never report any stale modules. Users upgrading to this version or later
versions would then see unexpected module load errors.
Since there are few users who actually want to install Asterisk into paths
that contain spaces, and a proper fix for the build system would take many hours,
the best solution for now is to just revert the partial solution.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
MeetMe requests a PIN twice in some circumstances
If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
options, MeetMe will ask for the PIN two times: once for creating the
conference and once for entering the conference. This behavior was introduced
in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
controlling PIN entry for joining a conference.
(closes AST-601)
Review: https://reviewboard.asterisk.org/r/1305/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
app_dial may double free a channel datastore
When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917)
Reported by: Mark Murawski
Tested by: Mark Murawski
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
........
r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines
Preserve sample rate quality of wideband mixmonitor recordings.
MixMonitor has the ability to record in any file format Asterisk supports,
but the quality of wideband audio is not preserved. This is because
regardless of the sample rate the call is being recorded in, the audio
is always downsampled to 8khz and then upsampled to whatever wideband
format it is being written as. This patch resolves this by requesting
the audio from the audiohook in the signed linear format closest to the
sample rate of the format we are writing. This fix is only possible for
Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
audio.
Review: https://reviewboard.asterisk.org/r/1314/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
