Tilghman Lesher [Wed, 23 Dec 2009 02:52:30 +0000 (02:52 +0000)]
Actually use tmp for something (brings trunk back into sync with 1.6 branches).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236183
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David Vossel [Tue, 22 Dec 2009 21:53:46 +0000 (21:53 +0000)]
fixes iax "can't compress subclass
4294967295" error
(closes issue #16456)
Reported by: dvossel
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236144
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David Vossel [Tue, 22 Dec 2009 17:00:08 +0000 (17:00 +0000)]
Merged revisions 236062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines
fixes issue with p->method incorrectly set to ACK
It is possible for a second ACK to come in for a retransmitted message.
If an ack does not match an unacked message in our queue, restore the previous
p->method as this ACK is completely ignored.
(closes issue #16295)
Reported by: omolenkamp
Patches:
issue16295_v2.diff uploaded by dvossel (license 671)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236063
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David Vossel [Tue, 22 Dec 2009 16:11:47 +0000 (16:11 +0000)]
update CHANGES to reflect the addition of the test framework
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236028
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David Vossel [Tue, 22 Dec 2009 16:09:11 +0000 (16:09 +0000)]
Unit Test Framework API
The Unit Test Framework is a new API that manages registration and
execution of unit tests in Asterisk with the purpose of verifying the
operation of C functions. The Framework consists of a single test
manager accompanied by a list of registered test functions defined
within the code. A test is defined, registered, and unregistered
from the framework using a set of macros which allow the test code
to only be compiled within asterisk when the TEST_FRAMEWORK flag is
enabled in menuselect. This allows the test code to exist in the
same file as the C functions it intends to verify. Registered tests
may be viewed and executed via a set of new CLI commands. CLI commands
are also present for generating and exporting test results into xml
and txt formats.
For more information and use cases please refer to the documentation
provided at the beginning of the test.h file.
Review: https://reviewboard.asterisk.org/r/447/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236027
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Jeff Peeler [Mon, 21 Dec 2009 19:54:20 +0000 (19:54 +0000)]
Merged revisions 235940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines
Change Monitor to not assume file to write to does not contain pathing.
227944 changed the fname_base argument to always append the configured monitor
path. This change was necessary to properly compare files for uniqueness.
If a full path is given though, nothing needs to be appended and that is
handled correctly now.
(closes issue #16377)
(closes issue #16376)
Reported by: bcnit
Patches:
res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235941
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Kevin P. Fleming [Mon, 21 Dec 2009 18:51:17 +0000 (18:51 +0000)]
Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235904
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Tilghman Lesher [Mon, 21 Dec 2009 17:00:46 +0000 (17:00 +0000)]
Merged revisions 235821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) | 8 lines
Send parking lot announcement to the channel which parked the call, not the park-ee.
(closes issue #16234)
Reported by: yeshuawatso
Patches:
20091210__issue16234.diff.txt uploaded by tilghman (license 14)
20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: yeshuawatso
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235822
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Alec L Davis [Sun, 20 Dec 2009 08:22:35 +0000 (08:22 +0000)]
restarts busydetector (if enabled) when DTMF is received after call is bridged.
(closes issue 0016389)
Reported by: alecdavis
Tested by: alecdavis
Patch
dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235774
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Alec L Davis [Sat, 19 Dec 2009 08:59:31 +0000 (08:59 +0000)]
app_dial optional parameter to option 'r' to allow play indication from indications.conf
(closes issue #14504)
Reported by: alecdavis
Tested by: alecdavis,jsmith
Patch
app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235740
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Jeff Peeler [Fri, 18 Dec 2009 22:51:37 +0000 (22:51 +0000)]
Merged revisions 235635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines
Correct CDR dispositions for BUSY/FAILED
This patch is simple in that it reorders the disposition defines so that the fix
for issue 12946 works properly (the default CDR disposition was changed to
AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to
ensure all CDR records are written.
The side effects of CDR changes are scary, so I'm documenting the test cases
performed to attempt to catch any regressions. The following tests were all
performed using 1.4 rev 195881 vs head (235571) + patch:
A calls B
C calls B (busy)
Hangup C
Hangup A
(Both SIP and features)
A calls B
A blind transfers to C
Hangup C
(Both SIP and features)
A calls B
A attended transfers to C
Hangup C
A calls B
A attended transfers to C (SIP)
C blind transfers to A (features)
Hangup A
All of the test scenario CDRs matched.
The following tests were performed just with the patch to ensure proper operation
(with unanswered=yes):
exten =>s,1,Answer
exten =>s,n,ResetCDR(w)
exten =>s,n,ResetCDR(w)
exten =>s,1,ResetCDR(w)
exten =>s,n,ResetCDR(w)
(closes issue #16180)
Reported by: aatef
Patches:
bug16180.patch uploaded by jpeeler (license 325)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235660
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Tilghman Lesher [Fri, 18 Dec 2009 22:40:46 +0000 (22:40 +0000)]
Merged revisions 235652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 Dec 2009) | 2 lines
Revise verbiage, per #asterisk-dev discussion
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Tilghman Lesher [Fri, 18 Dec 2009 21:19:43 +0000 (21:19 +0000)]
Merged revisions 235572 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines
Point to the typical missing package, not the cryptic "termcap support".
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235573
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Joshua Colp [Thu, 17 Dec 2009 23:21:07 +0000 (23:21 +0000)]
Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have
already happened when we detected the CNG tone so this was basically a noop.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235521
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Tilghman Lesher [Thu, 17 Dec 2009 17:19:08 +0000 (17:19 +0000)]
Merged revisions 235421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009) | 8 lines
Use context from which Macro is executed, not macro context, if applicable.
Also, ensure that the extension COULD match, not just that it won't match more.
(closes issue #16113)
Reported by: OrNix
Patches:
20091216__issue16113.diff.txt uploaded by tilghman (license 14)
Tested by: OrNix
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235422
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Jeff Peeler [Thu, 17 Dec 2009 00:52:03 +0000 (00:52 +0000)]
Fix call forwarding for analog phones.
(closes issue #16440)
Reported by: mmichelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235382
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Jeff Peeler [Wed, 16 Dec 2009 20:25:27 +0000 (20:25 +0000)]
Add auth_policy option to jabber.conf for auto user registration.
The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342
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Jared Smith [Wed, 16 Dec 2009 05:24:58 +0000 (05:24 +0000)]
Merged revisions 235181 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines
Add a line showing that we can use CIDR notation.
patch by jsmith, after discussion with jtodd
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235298
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Jeff Peeler [Wed, 16 Dec 2009 00:31:53 +0000 (00:31 +0000)]
Enhance AMI redirect to allow channels to be redirected to different places.
New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added
to redirect the second channel to a different location. Previously, it was only
possible to redirect both channels to the same place.
(closes issue #15853)
Reported by: haakon
Patches:
trunk-manager.c.patch uploaded by haakon (license 880)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235265
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Tilghman Lesher [Tue, 15 Dec 2009 23:51:05 +0000 (23:51 +0000)]
Is it Friday yet?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235229
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Jeff Peeler [Tue, 15 Dec 2009 23:41:20 +0000 (23:41 +0000)]
Change match criteria existence in ast_channel_cmp_cb to use ast_strlen_zero.
(closes issue #16161)
Reported by: may213
Patches:
core-show-channel.patch uploaded by may213 (license 454)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235226
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David Vossel [Tue, 15 Dec 2009 18:43:06 +0000 (18:43 +0000)]
reverse minor sip registration regression
A registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly. Origially
issue #14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved into
issue #15539. Both the tweak and the bug fix contained minor incorrect
logic that resulted in some SIP registrations to fail.
(issue #14331)
(issue #15539)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235132
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Tilghman Lesher [Tue, 15 Dec 2009 15:33:49 +0000 (15:33 +0000)]
Merged revisions 235052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009) | 4 lines
Mandatory argument checking
(closes issue #16446)
Reported by: nicchap
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235053
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Kevin P. Fleming [Tue, 15 Dec 2009 14:35:46 +0000 (14:35 +0000)]
spandsp does in fact support V.17 modulation at 14.4 kilobits per second,
so we should generate T38MaxBitRate of 14400 (even though that doesn't really
affect the FAX transmission much at all)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235010
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Alec L Davis [Tue, 15 Dec 2009 07:18:31 +0000 (07:18 +0000)]
Support option 'n', as applications like Playback, Background etc.
Suggested on asterisk-dev as trivial application change.
Reported by: alecdavis
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234976
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Alec L Davis [Tue, 15 Dec 2009 03:26:49 +0000 (03:26 +0000)]
Whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234940
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Alec L Davis [Tue, 15 Dec 2009 03:04:59 +0000 (03:04 +0000)]
restarts busydetector (if enabled) when DTMF is received.
(closes issue #16389)
Reported by: alecdavis
Tested by: alecdavis
Patch
dtmf_busydetector.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234897
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Alec L Davis [Tue, 15 Dec 2009 02:29:50 +0000 (02:29 +0000)]
fixes escape to extensions 'o' and 'a', for digits '0' and '*'
(closes issue #16437)
Reported by: alecdavis
Tested by: alecdavis
Patch
extension_o_a_fix.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234893
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Alec L Davis [Tue, 15 Dec 2009 00:54:44 +0000 (00:54 +0000)]
ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF.
(closes issue #16409)
Reported by: alecdavis
Tested by: alecdavis
Patch
bug_16409.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234855
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Tilghman Lesher [Mon, 14 Dec 2009 23:16:00 +0000 (23:16 +0000)]
Allow greetings-only mailboxes for Voicemail.
(closes issue #15132)
Reported by: floletarmo
Patches:
voicemail_changes.patch uploaded by floletarmo (license 784)
(with some additional changes by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234820
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Jason Parker [Mon, 14 Dec 2009 21:32:03 +0000 (21:32 +0000)]
Allow tonelist as argument to ReadExten.
ReadExten already supported playing a tonezone from indications.conf.
It now has the ability to use a tonelist like 440+480/2000|0/4000
(closes issue #15185)
Reported by: jcovert
Patches:
app_readexten.c.patch uploaded by jcovert (license 551)
Tested by: qwell
Patch modified by me, to maintain backwards compatibility.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234776
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Tilghman Lesher [Mon, 14 Dec 2009 21:13:18 +0000 (21:13 +0000)]
Merged revisions 234699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009) | 5 lines
Deal with the situation where .flavor exists but .version does not.
Also make the script slightly more portable, in keeping with autoconf syntax.
(closes issue #14737)
Reported by: davidw
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234700
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Leif Madsen [Mon, 14 Dec 2009 17:19:58 +0000 (17:19 +0000)]
Update IMAP build documentation.
Update the IMAP build documentation to show how to build on 64-bit
platforms.
(issue #16433)
Reported by: shrift
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234631
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Sean Bright [Mon, 14 Dec 2009 16:08:09 +0000 (16:08 +0000)]
The default rate for 'timing test' is actually 50/sec, not 100/sec as advertised.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234572
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Olle Johansson [Mon, 14 Dec 2009 10:46:20 +0000 (10:46 +0000)]
Merged revisions 234492 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines
Stop sending 183's after call hangup.
There where still cases where the 183 keep-alive mechanism would not stop
sending 183's even though the Asterisk server had sent a final reply to
the invite.
EDVX-28
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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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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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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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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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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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