Tilghman Lesher [Sun, 13 Dec 2009 09:41:43 +0000 (09:41 +0000)]
Trim leading/trailing spaces from the filename, to deal with common user error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234458
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Jeff Peeler [Fri, 11 Dec 2009 23:17:09 +0000 (23:17 +0000)]
Merged revisions 234379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines
Fix talking detection status after conference user is muted.
This patch ensures that when a conference user is muted that the accompanying
AMI Meetme talking off event is sent. Also, the meetme list output is updated
to show the muted user as unmonitored.
(closes issue #16247)
Reported by: dimas
Patches:
v3-16247.patch uploaded by dimas (license 88)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234380
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Jason Parker [Thu, 10 Dec 2009 21:01:39 +0000 (21:01 +0000)]
Merged revisions 234255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234255 | qwell | 2009-12-10 14:58:09 -0600 (Thu, 10 Dec 2009) | 9 lines
Fix unselecting of menuselect options via GLOBAL_MAKEOPTS and USER_MAKEOPTS.
(closes issue #16296)
Reported by: abelbeck
Patches:
issue16296-
20091210.diff uploaded by qwell (license 4)
(abelbeck described a fix, which I expanded upon)
Tested by: abelbeck, qwell, lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234256
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Tilghman Lesher [Thu, 10 Dec 2009 18:56:23 +0000 (18:56 +0000)]
Missed a case that emits a WARNING where none is warranted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234210
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Jeff Peeler [Thu, 10 Dec 2009 17:31:23 +0000 (17:31 +0000)]
Add audio announcement option to app_page
As described in the CHANGES file:
* MeetMe has a new option 'G' to play an announcement before joining a
conference.
* Page has a new option 'A(x)' which will playback an announcement
simultaneously to all paged phones (and optionally excluding the caller's one
using the new option 'n') before the call is bridged.
To add the new option to meetme, the conference flag options had to be extended
to 64 bits.
(closes issue #14365)
Reported by: dferrer
Patches:
page_announce.patch uploaded by dferrer (license 525)
modified by me
Review: https://reviewboard.asterisk.org/r/188/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173
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Tilghman Lesher [Thu, 10 Dec 2009 16:24:26 +0000 (16:24 +0000)]
Merged revisions 234095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) | 9 lines
When we receive no response at all to our INVITE, allow the channel to be destroyed.
(closes issue #15627)
Reported by: falves11
Patches:
20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14)
20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: falves11
Review: https://reviewboard.asterisk.org/r/446/
(closes issue #15716)
Reported by: dant
(closes issue #16270)
Reported by: corruptor
(closes issue #15356)
Reported by: falves11
(issue #16382)
Reported by: lftsy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234129
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Russell Bryant [Wed, 9 Dec 2009 23:35:24 +0000 (23:35 +0000)]
Move an entry from CHANGES to UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234055
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Russell Bryant [Wed, 9 Dec 2009 23:30:48 +0000 (23:30 +0000)]
Move an entry from CHANGES that should be in UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234053
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Russell Bryant [Wed, 9 Dec 2009 23:26:50 +0000 (23:26 +0000)]
Provide a real description of LOCAL_PEEK().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234051
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Russell Bryant [Wed, 9 Dec 2009 23:20:49 +0000 (23:20 +0000)]
Remove a feature from CHANGES that was listed twice for 1.6.2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234028
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Russell Bryant [Wed, 9 Dec 2009 23:13:28 +0000 (23:13 +0000)]
Fix up the faxdetect entry in CHANGES.
This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X
versions. The description of the feature was also no longer accurate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234008
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Russell Bryant [Wed, 9 Dec 2009 22:15:39 +0000 (22:15 +0000)]
Remove an entry from CHANGES that is already in UPGRADE.txt (where it should be).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233967
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Russell Bryant [Wed, 9 Dec 2009 19:59:13 +0000 (19:59 +0000)]
Blocked revisions 233879 via svnmerge
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r233879 | russell | 2009-12-09 13:58:46 -0600 (Wed, 09 Dec 2009) | 2 lines
Fix breakage of the "module load <module>" CLI command.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233880
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Russell Bryant [Wed, 9 Dec 2009 15:15:02 +0000 (15:15 +0000)]
Blocked revisions 233782 via svnmerge
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r233782 | russell | 2009-12-09 09:14:21 -0600 (Wed, 09 Dec 2009) | 22 lines
Set a module load priority for format modules.
A recent change to app_voicemail made it such that the module now assumes that
all format modules are available while processing voicemail configuration.
However, when autoloading modules, it was possible that app_voicemail was
loaded before the format modules. Since format modules don't depend on
anything, set a module load priority on them to ensure that they get loaded
first when autoloading.
This version of the patch is specific to Asterisk 1.4 and 1.6.0. These versions
did not already support module load priority in the module API. This adds a
trivial version of this which is just a module flag to include it in a pass before
loading "everything".
Thanks to mmichelson for the review!
(closes issue #16412)
Reported by: jiddings
Tested by: russell
Review: https://reviewboard.asterisk.org/r/445/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233783
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Tilghman Lesher [Tue, 8 Dec 2009 18:40:19 +0000 (18:40 +0000)]
Typo pointed out on #asterisk-dev (by atis_work)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233732
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Tilghman Lesher [Tue, 8 Dec 2009 18:22:44 +0000 (18:22 +0000)]
Find another ref leak and change how we manage module references.
(closes issue #16388, closes issue #16279, closes issue #16390)
Reported by: parisioa
Patches:
20091208__issue16388.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman
Review: https://reviewboard.asterisk.org/r/442/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233718
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Russell Bryant [Tue, 8 Dec 2009 18:00:16 +0000 (18:00 +0000)]
Set a module load priority for format modules.
A recent change to app_voicemail made it such that the module now assumes that
all format modules are available while processing voicemail configuration.
However, when autoloading modules, it was possible that app_voicemail was
loaded before the format modules. Since format modules don't depend on
anything, set a module load priority on them to ensure that they get loaded
first when autoloading.
This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will
require a different approach since the module load priority functionality is
not present in the module API.
(issue #16412)
Reported by: jiddings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233692
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Atis Lezdins [Tue, 8 Dec 2009 00:09:34 +0000 (00:09 +0000)]
Blocked revisions 233618 via svnmerge
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r233618 | atis | 2009-12-08 02:02:43 +0200 (Tue, 08 Dec 2009) | 15 lines
Merged revisions 233577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines
Fix compatibility with valgrind 3.3 and older.
(noticed in issue #16388)
Reported by: parisioa
Patches:
valgrind.supp uloaded by atis (license 242)
Tested by: atis, parisioa
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233619
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David Vossel [Mon, 7 Dec 2009 23:28:51 +0000 (23:28 +0000)]
fixes incorrect logic in ast_uri_encode
issue #16299
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233611
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David Vossel [Mon, 7 Dec 2009 23:26:22 +0000 (23:26 +0000)]
Blocked revisions 233609 via svnmerge
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r233609 | dvossel | 2009-12-07 17:24:59 -0600 (Mon, 07 Dec 2009) | 8 lines
hex escape control and non 7-bit clean characters in uri_encode
In ast_uri_encode, non 7-bit clean characters were being hex escaped
correctly, but control characters were not.
(issue #16299)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233610
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Atis Lezdins [Mon, 7 Dec 2009 23:10:13 +0000 (23:10 +0000)]
Fix compatibility with valgrind 3.3 and older.
(noticed in issue #16388)
Reported by: parisioa
Patches:
valgrind.supp uloaded by atis (license 242)
Tested by: atis, parisioa
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233577
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David Ruggles [Mon, 7 Dec 2009 19:48:14 +0000 (19:48 +0000)]
Fix TCP Client interface
Fix a couple of very minor bugs that prevent the socket client from working. The wrong set of properties were used in one place and the size of the address variable isn't set if the host name is an ip address. Also includes a fix for a bug that was introduced previously.
(closes issue #16121)
Reported by: thedavidfactor
Tested by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/439/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233545
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David Vossel [Mon, 7 Dec 2009 18:08:46 +0000 (18:08 +0000)]
Merged revisions 233471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines
fixes missing Contact header angle brackets
(closes issue #16298)
Reported by: mgernoth
Patches:
reg_parse_issue_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233472
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Jeff Peeler [Mon, 7 Dec 2009 17:59:46 +0000 (17:59 +0000)]
Add applications JabberJoin, JabberLeave, JabberSendGroup for XMPP groupchat
(closes issue #14352)
Reported by: fiddur
Patches:
trunk-14352-2.diff uploaded by phsultan (license 73)
Tested by: fiddur
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233468
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Matthew Nicholson [Mon, 7 Dec 2009 16:14:42 +0000 (16:14 +0000)]
Do not reject SDP packets describing only non audio streams.
(closes issue #16387)
Reported by: zalex1953
Patches:
media-level-c-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, zalex1953
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233394
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Matthew Nicholson [Mon, 7 Dec 2009 16:13:29 +0000 (16:13 +0000)]
Blocked revisions 233392 via svnmerge
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r233392 | mnicholson | 2009-12-07 10:11:16 -0600 (Mon, 07 Dec 2009) | 7 lines
Allow SDP packets with only video session information.
(closes issue #16387)
Reported by: zalex1953
Tested by: mnicholson, zalex1953
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233393
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Tilghman Lesher [Sun, 6 Dec 2009 07:01:06 +0000 (07:01 +0000)]
Move implementation of closefrom(3) from app.c to strcompat.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233358
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David Vossel [Fri, 4 Dec 2009 21:54:44 +0000 (21:54 +0000)]
Merged revisions 233279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines
clarify requirecalltoken option in iax.sample.conf
(closes issue #16223)
Reported by: bklang
Patches:
clarify-iax-requirecalltoken.patch uploaded by bklang (license 919)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233280
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Tilghman Lesher [Fri, 4 Dec 2009 21:06:22 +0000 (21:06 +0000)]
Using the builtin function breaks OpenBSD 4.2
(closes issue #16395)
Reported by: jtodd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233239
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David Vossel [Fri, 4 Dec 2009 20:21:11 +0000 (20:21 +0000)]
update CHANGES file for .m3u support in Mp3Player application
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233235
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David Vossel [Fri, 4 Dec 2009 20:19:53 +0000 (20:19 +0000)]
.m3u support for Mp3Player app
(closes issue #14823)
Reported by: macli
Patches:
app_mp3.diff1 uploaded by macli (license )
Tested by: macli, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233234
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David Vossel [Fri, 4 Dec 2009 19:02:06 +0000 (19:02 +0000)]
update CHANGES for new queue option, penaltymemberslimit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233198
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David Vossel [Fri, 4 Dec 2009 18:55:21 +0000 (18:55 +0000)]
changes penaltymemberslimit to use scanf for config value parsing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233197
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David Vossel [Fri, 4 Dec 2009 18:48:31 +0000 (18:48 +0000)]
new queue option, penaltymemberslimit, disregards penalty on too few queue members when enabled
(closes issue #14559)
Reported by: fiddur
Patches:
trunk-199584-1.diff uploaded by fiddur (license 678)
Tested by: fiddur, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233196
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David Vossel [Fri, 4 Dec 2009 17:22:31 +0000 (17:22 +0000)]
Merged revisions 233116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines
document and rename strip_control() in app_voicemail
(closes issue #16291)
Reported by: wdoekes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233121
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Russell Bryant [Fri, 4 Dec 2009 17:18:22 +0000 (17:18 +0000)]
Merged revisions 233092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) | 7 lines
Only do frame payload check for HOLD frames.
This code was added for helping to debug the source of invalid HOLD frames.
However, a side effect of this is that it will incorrectly report errors for
frames that have an integer payload. Make the check for this block specific
to the HOLD frame case.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233100
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Matthias Nick [Fri, 4 Dec 2009 17:15:47 +0000 (17:15 +0000)]
Parse global variables or expressions in hint extensions
Parse global variables or expressions in hint extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
(closes issue #16166)
Reported by: rmudgett
Tested by: mnick, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233093
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Michiel van Baak [Fri, 4 Dec 2009 16:55:07 +0000 (16:55 +0000)]
Let's unlock the lines list after the AST_LIST_TRAVERSE instead of inside it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233089
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Michiel van Baak [Fri, 4 Dec 2009 16:18:57 +0000 (16:18 +0000)]
Only assign line and device in handle_transfer_button when we have a subchannel.
(closes issue #16040)
Reported by: ebroad
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233059
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Tilghman Lesher [Fri, 4 Dec 2009 16:08:58 +0000 (16:08 +0000)]
Update the mysql driver to always return NULL columns, as this is needed for the realtime API to work correctly.
(closes issue #16138)
Reported by: sohosys
Patches:
20091029__issue16138.diff.txt uploaded by tilghman (license 14)
Tested by: sohosys
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233050
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Matthias Nick [Fri, 4 Dec 2009 15:38:33 +0000 (15:38 +0000)]
Merged revisions 233014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines
Warning message gets displayed only once
Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second.
(closes issue #15769)
Reported by: falves11
Patches:
patch_15769_14.txt uploaded by mnick (license 874)
Tested by: mnick, falves11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233046
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Tilghman Lesher [Fri, 4 Dec 2009 05:26:54 +0000 (05:26 +0000)]
Buildbot complained
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232982
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Tilghman Lesher [Fri, 4 Dec 2009 04:52:24 +0000 (04:52 +0000)]
OS X does not define MSG_NOSIGNAL, but it does have a socket option SO_NOSIGPIPE.
(closes issue #16178)
Reported by: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232950
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Tilghman Lesher [Thu, 3 Dec 2009 22:13:56 +0000 (22:13 +0000)]
Add pagerdateformat, to allow shorter dates for SMS messages.
(closes issue #16263)
Reported by: andrew
Patches:
pagerdate.patch uploaded by andrew (license 240)
(with a slight modification by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232916
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Tilghman Lesher [Thu, 3 Dec 2009 20:47:07 +0000 (20:47 +0000)]
Merged revisions 232820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines
Deprecate "cz" in favor of "cs".
Also, change the use of language codes so that language registers as a prefix,
rather than an exact match.
(closes issue #16272)
Reported by: patrol-cz
Patches:
20091203__issue16272.diff.txt uploaded by tilghman (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232854
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Alexandr Anikin [Thu, 3 Dec 2009 20:26:55 +0000 (20:26 +0000)]
jitterbuffer setup correction
correction of double pointer references from previous rev
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232853
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TransNexus OSP Development [Thu, 3 Dec 2009 08:47:38 +0000 (08:47 +0000)]
Replaced two deprecated functions of OSP Toolkit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232771
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TransNexus OSP Development [Thu, 3 Dec 2009 03:56:14 +0000 (03:56 +0000)]
Added custom info support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232738
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Jeff Peeler [Thu, 3 Dec 2009 00:38:03 +0000 (00:38 +0000)]
Extend voicemail to allow IMAP folders to be specified per mailbox.
Previously only possible per context, new option called imapfolder.
(closes issue #14298)
Reported by: jablko
Patches:
patch-
200906202 uploaded by jablko (license 675)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232700
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Tilghman Lesher [Thu, 3 Dec 2009 00:09:36 +0000 (00:09 +0000)]
Remove debugging line
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232661
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Tilghman Lesher [Thu, 3 Dec 2009 00:08:55 +0000 (00:08 +0000)]
Fix multiple issues with musiconhold, which led to classes not getting destroyed properly.
* Classes are now tracked past removal from the core container, and module
removal is actively prevented until all references are freed.
* A hanging reference stored in the channel has been removed. This could have
caused a mismatch and the music state not properly cleared, if two or more
reloads occurred between MOH being stopped and MOH being restarted.
* In certain circumstances, duplicate classes were possible.
* A race existed at reload time between a process being killed and the thread
responsible for reading from the related pipe respawning that process.
* Several reference counts have also been corrected. At least one could have
caused deleted classes to stick around forever, consuming resources. This
originally manifested as MOH external processes that were not killed at
reload time.
(closes issue #16279, closes issue #16207)
Reported by: parisioa, dcabot
Patches:
20091202__issue16279__2.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232660
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David Vossel [Wed, 2 Dec 2009 23:27:45 +0000 (23:27 +0000)]
update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 and 1.6.2
(closes issue #16212)
Reported by: miki
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232657
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David Ruggles [Wed, 2 Dec 2009 22:17:22 +0000 (22:17 +0000)]
Prevent double closing of FDs by EIVR
This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications.
EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance
the second close would then close the FD now in use by AGI.
(closes issue #16305)
Reported by: diLLec
Tested by: thedavidfactor, diLLec
Review: https://reviewboard.asterisk.org/r/436/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232587
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Jeff Peeler [Wed, 2 Dec 2009 22:02:43 +0000 (22:02 +0000)]
Merged revisions 232581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) | 7 lines
Send ack (response/message) after receiving manager action userevent
(closes issue #16264)
Reported by: dimas
Patches:
event-ack.patch uploaded by dimas (license 88)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232582
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Matthew Nicholson [Wed, 2 Dec 2009 21:37:18 +0000 (21:37 +0000)]
Fix support for multiline SMS messages in chan_mobile.
(closes issue #16278)
Reported by: Artem
Patches:
     multiline-sms-fix2.diff uploaded by mnicholson (license 96)
Tested by: Artem
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232580
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Jeff Peeler [Wed, 2 Dec 2009 21:32:50 +0000 (21:32 +0000)]
Make manager response to "Action: events" finish with empty line
(closes issue #16275)
Reported by: vnovy
Patches:
manager.c.diff uploaded by vnovy (license 922)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232576
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Matthew Nicholson [Wed, 2 Dec 2009 21:13:02 +0000 (21:13 +0000)]
Do something with the service indicator so that asterisk does not attempt to use a chan_mobile endpoint that does not have service.
(closes issue #16132)
Reported by: nikkk
Patches:
     service-indicator2.diff uploaded by mnicholson (license 96)
Tested by: nikkk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232544
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Joshua Colp [Wed, 2 Dec 2009 20:10:07 +0000 (20:10 +0000)]
Add an 'X' option to the asterisk application which enables #exec for configuration files.
This option can be used to enable #exec support in the asterisk.conf configuration file.
(closes issue #16260)
Reported by: atis
Patches:
exec_includes.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232510
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David Vossel [Wed, 2 Dec 2009 19:03:55 +0000 (19:03 +0000)]
Blocked revisions 232444 via svnmerge
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r232444 | dvossel | 2009-12-02 13:03:01 -0600 (Wed, 02 Dec 2009) | 9 lines
fixes app_queue ao2 error
(closes issue #16369)
Reported by: vrban
Patches:
queue_issue_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232445
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Joshua Colp [Wed, 2 Dec 2009 18:35:47 +0000 (18:35 +0000)]
Add an option to Record which enables a mode where any DTMF digit will terminate recording.
(closes issue #15436)
Reported by: Vince
Patches:
app_record.diff uploaded by Vince (license 823)
Tested by: dbrooks
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232442
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Mark Michelson [Wed, 2 Dec 2009 17:18:07 +0000 (17:18 +0000)]
Do not change the exten string field or rebuild the contact header
on an inbound sip_pvt if the outbound call is redirected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232365
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Joshua Colp [Wed, 2 Dec 2009 17:06:54 +0000 (17:06 +0000)]
Merged revisions 232355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines
Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239)
Reported by: CGMChris
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232356
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David Vossel [Wed, 2 Dec 2009 17:00:15 +0000 (17:00 +0000)]
Merged revisions 232350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) | 6 lines
ast_outaddrfor doesn't do htons() on port, looks odd in strace.
(closes issue #16290)
Reported by: wdoekes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232351
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Joshua Colp [Wed, 2 Dec 2009 16:40:14 +0000 (16:40 +0000)]
Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response.
(closes issue #16186)
Reported by: atis
Patches:
sip_t38_response_415.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232345
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David Vossel [Wed, 2 Dec 2009 15:42:54 +0000 (15:42 +0000)]
Merged revisions 232268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines
fixes segfault in func_groupcount
closes issue #16337)
Reported by: Parantido
Patches:
issue_16337.diff uploaded by dvossel (license 671)
Tested by: Parantido, dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232269
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Joshua Colp [Wed, 2 Dec 2009 14:54:28 +0000 (14:54 +0000)]
Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario
causing code to execute during reload that should not.
(issue AST-263)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232230
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Terry Wilson [Wed, 2 Dec 2009 04:06:14 +0000 (04:06 +0000)]
Blocked revisions 232165 via svnmerge
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r232165 | twilson | 2009-12-01 22:05:09 -0600 (Tue, 01 Dec 2009) | 2 lines
Fix compiling without devmode
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232166
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Tilghman Lesher [Wed, 2 Dec 2009 03:26:16 +0000 (03:26 +0000)]
So apparently, some platforms don't have ffsll(3).
The manpage lies; it says that the function is in POSIX, but that's only for
ffs(3), not ffsll(3).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232164
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Jeff Peeler [Wed, 2 Dec 2009 00:45:18 +0000 (00:45 +0000)]
Merged revisions 232090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines
Do not modify the gain settings on data calls.
(The digital flag actually represents a data call.)
(closes issue #15972)
Reported by: udosw
Patches:
transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232091
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Russell Bryant [Tue, 1 Dec 2009 23:56:14 +0000 (23:56 +0000)]
Use __builtin_ffsll() from gcc instead of ffssll() to fix a FreeBSD build error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232017
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Russell Bryant [Tue, 1 Dec 2009 23:38:34 +0000 (23:38 +0000)]
Fix a build error on FreeBSD.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232012
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Russell Bryant [Tue, 1 Dec 2009 23:27:53 +0000 (23:27 +0000)]
Merged revisions 232007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines
Fix a warning pointed out by buildbot.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232008
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Jeff Peeler [Tue, 1 Dec 2009 21:54:21 +0000 (21:54 +0000)]
Merged revisions 231911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines
Fix crash with invalid frame data
The crash was happening as a result of a frame containing an invalid data
pointer, but was set with data length of zero. The few times the issue was
reproduced it _seemed_ that the frame was queued properly, that is the data
pointer was set to NULL. I never could reproduce the crash so as a last resort
the crash has been fixed, but a check in __ast_read has been added to give as
much information about the source of problematic frames in the future.
(closes issue #16058)
Reported by: atis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231927
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David Vossel [Tue, 1 Dec 2009 21:20:19 +0000 (21:20 +0000)]
Merged revisions 231853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) | 3 lines
WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231867
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Tilghman Lesher [Tue, 1 Dec 2009 20:27:37 +0000 (20:27 +0000)]
More 32->64 bit codec conversions.
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850
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Tilghman Lesher [Tue, 1 Dec 2009 17:48:54 +0000 (17:48 +0000)]
Formats need to be able to represent all 64 codec bits.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231814
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Matthew Nicholson [Tue, 1 Dec 2009 15:47:36 +0000 (15:47 +0000)]
Merged revisions 231740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines
Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231741
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Kevin P. Fleming [Mon, 30 Nov 2009 21:47:42 +0000 (21:47 +0000)]
Another round of UDPTL stack fixes/improvements:
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231692
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Matthew Nicholson [Mon, 30 Nov 2009 21:31:55 +0000 (21:31 +0000)]
Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688
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Matthew Nicholson [Mon, 30 Nov 2009 21:21:29 +0000 (21:21 +0000)]
Reverted 231616
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231637
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Matthew Nicholson [Mon, 30 Nov 2009 21:13:42 +0000 (21:13 +0000)]
Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231616
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Joshua Colp [Mon, 30 Nov 2009 20:44:30 +0000 (20:44 +0000)]
When receiving SDP that matches the version of the last one do not treat it as a fatal error.
(closes issue #16238)
Reported by: seandarcy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231602
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David Vossel [Mon, 30 Nov 2009 18:55:07 +0000 (18:55 +0000)]
app_queue crashes randomly, often during call-transfers
This patch adds a ref to the queue_ent object's parent call_queue
in queue_exec() so the call_queue won't be destroyed
while the the queue_ent still holds a pointer to it.
(closes issue 0015686)
Tested by: dvossel, aragon
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231556
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David Vossel [Mon, 30 Nov 2009 17:28:28 +0000 (17:28 +0000)]
Merged revisions 231441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) | 11 lines
fixes crash caused by RTP comfort noise payload greater than 24 bytes
AST-2009-010
(closes issue #16242)
Reported by: amorsen
Patches:
issue16242.diff uploaded by oej (license 306)
Tested by: amorsen, oej, dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231491
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Tilghman Lesher [Mon, 30 Nov 2009 16:53:10 +0000 (16:53 +0000)]
Export dynamic (weak-linked) symbols correctly.
(closes issue #15193)
Reported by: eliel
Patches:
20091111__issue15193.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231439
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David Vossel [Mon, 30 Nov 2009 16:37:33 +0000 (16:37 +0000)]
Blocked revisions 231437 via svnmerge
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r231437 | dvossel | 2009-11-30 10:32:58 -0600 (Mon, 30 Nov 2009) | 18 lines
app_queue crashes randomly, often during call-transfers
In app_queue, it is possible for a call_queue to be destroyed
while another object still holds a pointer to it. This patch
converts call_queue objects to ao2 objects allowing them to be
ref counted. This makes it safe for the queue_ent object in
queue_exec() to reference it's parent call_queue even after it
has left the queue.
(closes issue #15686)
Reported by: Hatrix
Patches:
v2_queue_ao2.diff uploaded by dvossel (license 671)
Tested by: dvossel, aragon
Review: https://reviewboard.asterisk.org/r/427/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231438
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Joshua Colp [Mon, 30 Nov 2009 16:29:29 +0000 (16:29 +0000)]
Fix a bug where an immediate masquerade would cause a queued unhold frame to get lost. Now we just
indicate unhold directly after the masquerade is complete.
(issue ABE-2011)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231436
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TransNexus OSP Development [Fri, 27 Nov 2009 08:47:18 +0000 (08:47 +0000)]
1. Modified exported variable names.
2. Added destination port support.
3. Added new protocols.
4. Added QoS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231401
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Tilghman Lesher [Thu, 26 Nov 2009 02:09:58 +0000 (02:09 +0000)]
Reorder option flags. Change guidelines so that example code is consistent with guidelines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231369
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Tilghman Lesher [Wed, 25 Nov 2009 22:33:02 +0000 (22:33 +0000)]
Merged revisions 231298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines
After a frame duplication failure, unlock the channel before returning.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231299
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David Vossel [Wed, 25 Nov 2009 21:39:00 +0000 (21:39 +0000)]
Blocked revisions 231235 via svnmerge
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r231235 | dvossel | 2009-11-25 15:38:32 -0600 (Wed, 25 Nov 2009) | 9 lines
fixes solaris segfault on dial with verbosity >= 3
(closes issue #16193)
Reported by: asgaroth
Patches:
bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231236
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David Vossel [Wed, 25 Nov 2009 21:24:31 +0000 (21:24 +0000)]
Blocked revisions 231233 via svnmerge
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r231233 | dvossel | 2009-11-25 15:23:41 -0600 (Wed, 25 Nov 2009) | 8 lines
fixes conditional jump or move depending on uninitialised STACK value
(closes issue #16261)
Reported by: edguy3
Patches:
edguy16261.patch uploaded by edguy3 (license 917)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231234
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Matthew Nicholson [Wed, 25 Nov 2009 15:42:48 +0000 (15:42 +0000)]
Load pbx_lua with global symbols to allow linking with other lua libraries.
Found by Maxim Litnitskiy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231189
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Tilghman Lesher [Tue, 24 Nov 2009 20:31:28 +0000 (20:31 +0000)]
Found a few places where queue refcounts were counted incorrectly. Also add debug statements.
(closes issue #15982, closes issue #15984)
Reported by: atis
Patches:
20091111__issue15982.diff.txt uploaded by tilghman (license 14)
Tested by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231134
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Jeff Peeler [Tue, 24 Nov 2009 18:50:36 +0000 (18:50 +0000)]
Fix erroneous hangup extension execution
ast_spawn_extension behaves differently from 1.4 in that hangups and extensions
that do not exist do not return an error, whereas in 1.6 it does. This is now
taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set
properly.
(closes issue #16106)
Reported by: ajohnson
Tested by: ajohnson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231095
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Jeff Peeler [Tue, 24 Nov 2009 17:12:25 +0000 (17:12 +0000)]
Fix problem on digital channels due to digital flag not getting set
Changed areas in sig_pri to set the digital flag using a callback that will
also set the corresponding flag in chan_dahdi. Modified dahdi_request slightly
so that if a bearer is marked as digital, that information is available when
creating the new channel.
(closes issue #16151)
Reported by: alecdavis
Patch based on bug_16151.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231058
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Matthew Nicholson [Tue, 24 Nov 2009 13:52:21 +0000 (13:52 +0000)]
Updated CHANGES file to describe the new 'd' option to app_followme added in r230964
(related to issue #14155)
Reported by: junky
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231025
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Tilghman Lesher [Tue, 24 Nov 2009 04:58:44 +0000 (04:58 +0000)]
Add REPLACE & PASSTHRU functions, overhaul of func_strings, fix API docs for the ast_get_encoded_* functions.
* Add REPLACE function, which searches a given variable for a set of
characters and replaces each with a given character.
* Add PASSTHRU function, which passes a literal string back, like a NoOp for
functions. Intent is to be able to specify a literal string to another
function that takes a variable name as an argument.
* Let the array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation functions to
modify ASTDB and ODBC backends, assuming the func_odbc configuration has
both read and write functions.
(closes issue #15223)
Reported by: ajohnson
Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230994
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Matthew Nicholson [Mon, 23 Nov 2009 22:37:39 +0000 (22:37 +0000)]
Add an option to app_followme to disable the "please hold" announcement.
(closes issue #14155)
Reported by: junky
Patches:
M14555-trunk.diff uploaded by junky (license 177) (modified)
Tested by: junky
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230964
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Joshua Colp [Mon, 23 Nov 2009 15:45:45 +0000 (15:45 +0000)]
Change fax detection in chan_sip so it behaves as one would expect.
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230881
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