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git.ipfire.org Git - thirdparty/asterisk.git/log
Terry Wilson [Wed, 13 Oct 2010 22:58:43 +0000 (22:58 +0000)]
Merged revisions 291577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines
Don't ignore frames that have been queued when softhangup'd
When an outgoing call is answered and hung up by the far end *very* quickly, we
may not read any frames and therefor end up with a call that displays the wrong
disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately
sets the _softhangup flag on the channel and then queues the HANGUP control
frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates
that a hangup request has been made (which it will if _softhangup is set). So,
we end up losing control frames.
This change makes __ast_read continue to read frames even if a soft hangup has
been requested. It queues a hangup frame to make sure that __ast_read() will
still eventually return NULL.
Much thanks to David Vossel for all of the reviews, discussion, and help!
(closes issue #16946)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/740/
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Russell Bryant [Wed, 13 Oct 2010 15:29:21 +0000 (15:29 +0000)]
Merged revisions 291392 via svnmerge from
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r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines
Lock pvt so pvt->owner can't disappear when queueing up a frame.
This fixes a crash due to a hangup race condition.
ABE-2601
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Leif Madsen [Tue, 12 Oct 2010 17:20:02 +0000 (17:20 +0000)]
Add undocumented variables to phoneprov.conf.sample
(closes issue #18107)
Reported by: lathama
Patches:
phoneprov.conf.sample.diff uploaded by lathama (license 1028)
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Tilghman Lesher [Tue, 12 Oct 2010 17:05:31 +0000 (17:05 +0000)]
Merged revisions 291263 via svnmerge from
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r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines
Oops, incorrect range (although unallocated at ARIN)
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Leif Madsen [Tue, 12 Oct 2010 16:07:28 +0000 (16:07 +0000)]
Add documention that mentions options are defined but not used.
(Issue #18101)
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Richard Mudgett [Mon, 11 Oct 2010 18:39:06 +0000 (18:39 +0000)]
Make exit from handle_request_do() consistent.
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Richard Mudgett [Mon, 11 Oct 2010 18:34:22 +0000 (18:34 +0000)]
Merged revisions 291109 via svnmerge from
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r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line
Add missing unlock to an exception condition in reload_config().
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Richard Mudgett [Mon, 11 Oct 2010 16:39:17 +0000 (16:39 +0000)]
Fixed infinite loop in verbose/debug message output.
Setting the module/filename specific message level and then changing it
resulted in the linked list being looped on itself. Traversing this
linked list is an infinite loop if what you are looking for is not in the
list.
Also plugged some CLI parsing holes in the associated CLI command:
* Removing a nonexistent module from the list actually added it with a
level of zero.
* Setting the non-module specific level to zero is now equivalent to
setting it to "off" as documented.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@291073
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Jeff Peeler [Fri, 8 Oct 2010 02:45:44 +0000 (02:45 +0000)]
Merged revisions 290862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines
Ensure editline cleanup occurs when Ctrl-C is pressed at control console.
A recent change was made to avoid a race condition on shutdown which only called
the end functions from the console thread. However, when pressing Ctrl-C the
quit handler is called from the signal handler thread.
(closes issue #17698)
Reported by: jmls
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Jason Parker [Thu, 7 Oct 2010 20:57:14 +0000 (20:57 +0000)]
Merged revisions 290750 via svnmerge from
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r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines
Allow PRI to build properly when using --with-pri.
Use the directories found for the parent when using lib dependencies.
(closes issue #17314)
Reported by: tzafrir
Patches:
17314-withdeps.diff uploaded by qwell (license 4)
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Russell Bryant [Thu, 7 Oct 2010 10:53:56 +0000 (10:53 +0000)]
Don't crash when Set() is called without a value.
Review: https://reviewboard.asterisk.org/r/949/
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Tilghman Lesher [Wed, 6 Oct 2010 13:48:27 +0000 (13:48 +0000)]
Allow streaming audio from a pipe.
(closes issue #18001)
Reported by: jamicque
Patches:
20100926__issue18001.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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Tilghman Lesher [Tue, 5 Oct 2010 20:21:02 +0000 (20:21 +0000)]
Merged revisions 290392 via svnmerge from
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r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) | 8 lines
Fix a crash by ensuring that we don't alter memory after it's freed.
(closes issue #17387)
Reported by: jmls
Patches:
20100726__issue17387.diff.txt uploaded by tilghman (license 14)
Tested by: jmls
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David Vossel [Tue, 5 Oct 2010 19:54:50 +0000 (19:54 +0000)]
Fixes PickupChan() not working with full channel name.
(closes issue #18011)
Reported by: schern
Patches:
app_directed_pickup.c.2.patch uploaded by schern (license 995)
app_directed_pickup.c.trunk.patch uploaded by schern (license 995)
Tested by: schern, dvossel
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Richard Mudgett [Tue, 5 Oct 2010 17:42:02 +0000 (17:42 +0000)]
Merged revisions 290323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r290323 | rmudgett | 2010-10-05 12:41:18 -0500 (Tue, 05 Oct 2010) | 11 lines
Merged revision 258974 from
https://origsvn.digium.com/svn/asterisk/trunk
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r258974 | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4 lines
Line 24 missed in compatibility fix in revision 233577
added a "fun:" prefix line 24
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Tilghman Lesher [Mon, 4 Oct 2010 23:14:59 +0000 (23:14 +0000)]
Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored.
Also change the AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this behavior.
(closes issue #17366)
Reported by: murf
Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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Tilghman Lesher [Mon, 4 Oct 2010 20:22:03 +0000 (20:22 +0000)]
Merged revisions 290177 via svnmerge from
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r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 Oct 2010) | 2 lines
Fixing Mac OS X auto-builder.
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Tilghman Lesher [Sun, 3 Oct 2010 21:06:58 +0000 (21:06 +0000)]
Merged revisions 290100 via svnmerge from
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r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 Oct 2010) | 2 lines
Automatically re-run configure test for menuselect, when the relevant makeopts settings change.
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Olle Johansson [Sat, 2 Oct 2010 08:52:03 +0000 (08:52 +0000)]
Merged revisions 289949 via svnmerge from
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r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines
Add documentation for undocumented option to AMI action originate
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Tilghman Lesher [Sat, 2 Oct 2010 04:45:49 +0000 (04:45 +0000)]
Merged revisions 289873 via svnmerge from
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r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines
When forwarding a message, a prepend means that the filesystem will always have a better copy.
(closes issue #17803)
Reported by: dpetersen
Patches:
20100923__issue17803.diff.txt uploaded by tilghman (license 14)
Tested by: dpetersen
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Jeff Peeler [Fri, 1 Oct 2010 23:01:31 +0000 (23:01 +0000)]
Merged revisions 289797 via svnmerge from
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r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
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Paul Belanger [Fri, 1 Oct 2010 17:09:03 +0000 (17:09 +0000)]
Merged revisions 289703 via svnmerge from
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r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct 2010) | 6 lines
Disable debugging by default
and reformat .config file.
Review: https://reviewboard.asterisk.org/r/929/
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Jeff Peeler [Fri, 1 Oct 2010 16:21:04 +0000 (16:21 +0000)]
Merged revisions 289699 via svnmerge from
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r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines
Ensure user portion of SIP URI matches dialplan when using encoded characters.
This commit takes a simliar approach to 288112 and checks the dialplan to
determine the proper action for an incoming contact header as to whether or not
it should be decoded or not. sip_new was blindly always decoding the extension,
which also caused the outgoing contact header to be incorrect as well as failing
to match the encoded extension in the dialplan.
(closes issue #17892)
Reported by: wdoekes
Patches:
bug17892-1.patch uploaded by jpeeler (license 325)
Tested by: wdoekes
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Stefan Schmidt [Fri, 1 Oct 2010 09:42:22 +0000 (09:42 +0000)]
don't iterate through all dialogs to find and delete old subscribes
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed.
(closes issue #17950)
Reported by: schmidts
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/901/
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Matthew Nicholson [Thu, 30 Sep 2010 19:51:27 +0000 (19:51 +0000)]
Properly handle channel allocation failures duing invites with replaces.
ABE-2588
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Brett Bryant [Thu, 30 Sep 2010 17:09:48 +0000 (17:09 +0000)]
Merged revisions 289500 via svnmerge from
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r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010) | 11 lines
res_agi.c:handle_getvariablefull() could recursively lock a channel and not
release it if an argument is the current channel's name.
(closes issue #17970)
Reported by: mdu113
Patches:
res_agi.c.diff3 uploaded by mdu113 (license 582)
Tested by: mdu113
Review: https://reviewboard.asterisk.org/r/947/
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Russell Bryant [Thu, 30 Sep 2010 15:37:29 +0000 (15:37 +0000)]
Merged revisions 289424 via svnmerge from
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r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines
Fix a crash in app_sms.
Since the data being passed to the generator callback is on the stack of the
SMS() application, we must ensure that the generator is stopped before the
application exits.
ABE-2587
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Jason Parker [Wed, 29 Sep 2010 21:03:47 +0000 (21:03 +0000)]
Merged revisions 289338 via svnmerge from
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r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines
Allow a manager originate to succeed on forwarded devices.
The timeout to wait for an answer was being set to 0 when a device forwarded to another
extension. We don't always need the timeout set like this, so make it an optional
parameter, and don't use it in this case.
ABE-2544
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Leif Madsen [Wed, 29 Sep 2010 20:24:47 +0000 (20:24 +0000)]
Update sample documentation to note md5secret requirements.
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Russell Bryant [Wed, 29 Sep 2010 20:15:57 +0000 (20:15 +0000)]
Don't completely ignore md5secret from LDAP if the value does not begin with {md5}.
This fixes a problem that lmadsen ran in to where md5secret was not working for him.
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Matthew Nicholson [Wed, 29 Sep 2010 15:04:11 +0000 (15:04 +0000)]
Merged revisions 289177 via svnmerge from
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r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines
Set the caller id on CDRs when it is set on the parent channel.
(closes issue #17569)
Reported by: tbelder
Patches:
17569.diff uploaded by tbelder (license 618)
Tested by: tbelder
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Brett Bryant [Tue, 28 Sep 2010 18:14:19 +0000 (18:14 +0000)]
Merged revisions 289094 via svnmerge from
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r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines
Fixes an issue with the Newchannel AMI event during the Masquerading process.
Fixes an issue with the Newchannel AMI event during the Masquerading process,
where no Newchannel AMI event was generated for the psuedo channel used during
the masquerading process.
(closes issue #17987)
Reported by: RadicAlish
Patches:
newchannel.patch.txt uploaded by RadicAlish (license 1122)
Tested by: RadicAlish
Review: https://reviewboard.asterisk.org/r/937/
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Terry Wilson [Fri, 24 Sep 2010 15:37:39 +0000 (15:37 +0000)]
Merged revisions 288746 via svnmerge from
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r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) | 5 lines
Don't fail a masquerade if it is already being hung up
This avoids noise on some Local channel situations where we don't use /n.
Thanks to Alec Davis for the suggestion.
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Tilghman Lesher [Fri, 24 Sep 2010 13:53:30 +0000 (13:53 +0000)]
Solaris won't printf a NULL.
(closes issue #18041)
Reported by: asgaroth
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Tilghman Lesher [Fri, 24 Sep 2010 03:36:01 +0000 (03:36 +0000)]
Merged revisions 288636 via svnmerge from
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r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines
Solaris compatibility fixes
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Terry Wilson [Wed, 22 Sep 2010 23:10:09 +0000 (23:10 +0000)]
Merged revisions 288499 via svnmerge from
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r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) | 8 lines
Don't let a Local channel get bridged to itself
If a local channel gets bridged to itself, it becomes orphaned with no devices
left to actually tell it to hang up. This patch modifies local_fixup() to detect
this case and deny it.
Review: https://reviewboard.asterisk.org/r/934
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David Vossel [Wed, 22 Sep 2010 17:49:05 +0000 (17:49 +0000)]
Merged revisions 288416 via svnmerge from
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r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines
RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response.
ABE-2458
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Matthew Nicholson [Wed, 22 Sep 2010 17:40:51 +0000 (17:40 +0000)]
Blocked revisions 288412 via svnmerge
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r288412 | mnicholson | 2010-09-22 12:39:12 -0500 (Wed, 22 Sep 2010) | 6 lines
Handle media specific T.38 SDP information
(closes issue #16647)
Reported by: kwemheuer
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David Vossel [Wed, 22 Sep 2010 16:53:28 +0000 (16:53 +0000)]
Merged revisions 288343 via svnmerge from
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r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines
During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup.
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Russell Bryant [Wed, 22 Sep 2010 16:44:13 +0000 (16:44 +0000)]
Merged revisions 288339 via svnmerge from
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r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines
Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf.
The handling of -c and console=yes should be the same, but they were not.
When you specify -c, it sets both a flag for console module and for asterisk
not to fork() off into the background. The handling of console=yes only set
console mode, so you would end up with a background process() trying to run
the Asterisk console and freaking out since it didn't have anything to read
input from.
Thanks to beagles for reporting and helping debug the problem!
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Tilghman Lesher [Wed, 22 Sep 2010 15:11:09 +0000 (15:11 +0000)]
Merged revisions 288265-288266 via svnmerge from
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r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) | 9 lines
Allow the encoding to be set, in case local charset does not agree with database.
(closes issue #16940)
Reported by: jamicque
Patches:
20100827__issue16940.diff.txt uploaded by tilghman (license 14)
20100921__issue16940__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) | 5 lines
Document addition of encoding parameter.
(issue #16940)
Reported by: jamicque
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Richard Mudgett [Wed, 22 Sep 2010 00:03:37 +0000 (00:03 +0000)]
Merged revisions 288192 via svnmerge from
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r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines
In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel.
Near the beginning of schedule_delivery(), ast_bridged_channel() is called
on iaxs[fr->callno]->owner. However, the channel is not locked, which can
result in ast_bridged_channel() crashing should owner->tech change to a
technology that doesn't implement bridged_channel.
I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since
the owner lock was not held there either.
Converted the existing channel deadlock avoidance to use
iax2_lock_owner(). Using the new function simplified some awkward code.
In the process of fixing the locking on ast_bridged_channel(), I also
found a memory leak in socket_process() for v1.6.2 and v1.8. The local
struct variable ies.vars is not freed on early/abnormal function exits.
(closes issue #17919)
Reported by: rain
Patches:
issue17919_v1.4.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/926/
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Paul Belanger [Tue, 21 Sep 2010 22:22:43 +0000 (22:22 +0000)]
Setup timer before set_config().
(closes issue #18019)
Reported by: Netview
Patches:
issue_0018019.patch uploaded by pabelanger (license 224)
Tested by: Netview
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@288147
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Tilghman Lesher [Tue, 21 Sep 2010 22:19:00 +0000 (22:19 +0000)]
Blocked revisions 288116 via svnmerge
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r288116 | tilghman | 2010-09-21 17:17:33 -0500 (Tue, 21 Sep 2010) | 2 lines
Underquoted definition of stringfield macro.
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Tilghman Lesher [Tue, 21 Sep 2010 21:59:46 +0000 (21:59 +0000)]
Merged revisions 288112 via svnmerge from
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r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines
Try both the encoded and unencoded subscription URI for a match in hints.
When a phone sends an encoded URI for a subscription, the URI is not matched
with the actual hint that is in decoded format. For example, if we have an
extension with a hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI, but when it's
decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the
correct hint.
(closes issue #17785)
Reported by: ramonpeek
Patches:
20100831__issue17785.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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Brett Bryant [Tue, 21 Sep 2010 19:46:20 +0000 (19:46 +0000)]
Merged revisions 288005 via svnmerge from
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r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines
Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate
memory on the first frame being queued in ast_queue_frame.
(closes issue #17882)
Reported by: seanbright
Tested by: seanbright
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Tilghman Lesher [Tue, 21 Sep 2010 19:07:53 +0000 (19:07 +0000)]
Merged revisions 287933 via svnmerge from
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r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) | 2 lines
Less than zero is an error, not any non-zero value.
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Brett Bryant [Mon, 20 Sep 2010 23:58:26 +0000 (23:58 +0000)]
Merged revisions 287758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines
Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag.
When using the 'a' MeetMe flag and having a user and admin pin setup for your
conference, using the user pin would gain you admin priviledges. Also, when no
user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the
user tried to enter a conference then they were still prompted for a pin and
forced to hit #.
(closes issue #17908)
Reported by: kuj
Patches:
pins_2.patch uploaded by kuj (license 1111)
Tested by: kuj
Review: [full review board URL with trailing slash]
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Alec L Davis [Mon, 20 Sep 2010 23:16:45 +0000 (23:16 +0000)]
ast_channel_masquerade: Avoid recursive masquerades.
Check all 4 combinations of (original/clonechan) * (masq/masqr).
Initially original->masq and clonechan->masqr were only checked.
It's possible with multiple masq's planned - and not yet executed, that
the 'original' chan could already have another masq'd into it - thus original->masqr
would be set, that masqr would lost.
Likewise for the clonechan->masq.
(closes issue #16057;#17363)
Reported by: amorsen;davidw,alecdavis
Patches:
based on bug16057.diff4.txt uploaded by alecdavis (license 585)
Tested by: ramonpeek, davidw, alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@287685
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Jason Parker [Mon, 20 Sep 2010 21:28:32 +0000 (21:28 +0000)]
Don't crash when parking a non-bridged call.
(closes issue #17680)
Reported by: jmhunter
Patches:
chan_skinny-park-v1.txt uploaded by DEA (license 3)
Tested by: jmhunter, DEA
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@287642
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Matthew Nicholson [Mon, 20 Sep 2010 15:56:21 +0000 (15:56 +0000)]
Use ast_str when processing hint state changes
Merged revisions 287555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines
Use ast_dynamic_str when processing hint state changes
(related to issue #17928)
Reported by: mdu113
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Matthew Nicholson [Mon, 20 Sep 2010 15:54:14 +0000 (15:54 +0000)]
Revert r287556.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@287557
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Matthew Nicholson [Mon, 20 Sep 2010 15:51:47 +0000 (15:51 +0000)]
Use ast_str when processing hint state changes
Merged revisions 287555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines
Use ast_dynamic_str when processing hint state changes
(related to issue #17928)
Reported by: mdu113
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Olle Johansson [Sun, 19 Sep 2010 16:06:10 +0000 (16:06 +0000)]
Merged revisions 287469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines
Make sure we always free variables properly in manager originate.
(closes issue #17891)
reported, solved and tested by oej
Review: https://reviewboard.asterisk.org/r/869/
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Tilghman Lesher [Fri, 17 Sep 2010 21:08:00 +0000 (21:08 +0000)]
Merged revisions 287386 via svnmerge from
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r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines
Blank columns should get set on reload, not ignored.
(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
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Matthew Nicholson [Fri, 17 Sep 2010 13:36:07 +0000 (13:36 +0000)]
Merged revisions 287307 via svnmerge from
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r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines
Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed().
(related to issue #17928)
Reported by: mdu113
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Jason Parker [Thu, 16 Sep 2010 22:12:51 +0000 (22:12 +0000)]
Merged revisions 287197 via svnmerge from
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r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) | 7 lines
Add LSB headers for Debian init script, since Debian will complain if it isn't there.
Headers were taken from trunk.
(closes issue #17958)
Reported by: javyer
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Matthew Nicholson [Thu, 16 Sep 2010 20:06:16 +0000 (20:06 +0000)]
Merged revisions 287118 via svnmerge from
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r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928)
Reported by: mdu113
Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113
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Matthew Nicholson [Thu, 16 Sep 2010 19:53:41 +0000 (19:53 +0000)]
Merged revisions 287114 via svnmerge from
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r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't stop printing cdr variables if we encounter one with a blank name or value.
(closes issue #17900)
Reported by: under
Patches:
core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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Jeff Peeler [Wed, 15 Sep 2010 20:28:02 +0000 (20:28 +0000)]
Merged revisions 286941 via svnmerge from
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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Matthew Nicholson [Tue, 14 Sep 2010 19:27:28 +0000 (19:27 +0000)]
Merged revisions 286756 via svnmerge from
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r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines
Don't clear the username from a realtime database when a registration expires.
Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either.
(closes issue #17551)
Reported by: ricardolandim
Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96)
Tested by: ricardolandim, mnicholson
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Matthew Nicholson [Tue, 14 Sep 2010 18:02:24 +0000 (18:02 +0000)]
Merged revisions 286679 via svnmerge from
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
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Tilghman Lesher [Tue, 14 Sep 2010 05:06:05 +0000 (05:06 +0000)]
Add documentation on missing backend tables for Voicemail
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286587
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Tilghman Lesher [Mon, 13 Sep 2010 23:48:51 +0000 (23:48 +0000)]
C precedence got me
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286557
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Tilghman Lesher [Mon, 13 Sep 2010 23:03:26 +0000 (23:03 +0000)]
Refactor conversion to ast_poll() to fix callparking regression.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286527
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Jason Parker [Mon, 13 Sep 2010 19:38:35 +0000 (19:38 +0000)]
Remove "Internal IP" from sip show settings, as it's not at all useful to display.
(closes issue #17840)
Reported by: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286456
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Jason Parker [Mon, 13 Sep 2010 15:13:24 +0000 (15:13 +0000)]
Blocked revisions 286381 via svnmerge
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r286381 | qwell | 2010-09-13 10:12:51 -0500 (Mon, 13 Sep 2010) | 5 lines
Add stuff to svn:ignore for tests/ directory.
(closes issue #17983)
Reported by: oej
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Olle Johansson [Sat, 11 Sep 2010 17:05:16 +0000 (17:05 +0000)]
Merged revisions 286267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/
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Terry Wilson [Fri, 10 Sep 2010 22:56:14 +0000 (22:56 +0000)]
Merged revisions 286222 via svnmerge from
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r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10 Sep 2010) | 1 line
Return -1 if chan_local doesn't support an option
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Paul Belanger [Fri, 10 Sep 2010 20:55:06 +0000 (20:55 +0000)]
Merged revisions 286114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines
Load iax.conf before registering any functions/applications/actions.
Review: https://reviewboard.asterisk.org/r/914/
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Richard Mudgett [Fri, 10 Sep 2010 20:42:44 +0000 (20:42 +0000)]
Merged revisions 286113 via svnmerge from
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r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
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Terry Wilson [Fri, 10 Sep 2010 20:35:25 +0000 (20:35 +0000)]
Merged revisions 286059 via svnmerge from
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r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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David Vossel [Fri, 10 Sep 2010 20:04:43 +0000 (20:04 +0000)]
Blocked revisions 286070 via svnmerge
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r286070 | dvossel | 2010-09-10 15:03:50 -0500 (Fri, 10 Sep 2010) | 32 lines
Fixes sip extension state update DEADLOCK
PROBLEM:
In chan_sip, and all the other channel drivers, it is common for
us to hold the tech_pvt lock while we ask the Asterisk core about
an extension and context. Every time we do this the locking
order becomes, (1. tech_pvt lock ---> 2. global context lock). In
chan_sip when a dialog subscribes to a hint, that locking order
is reversed in the extensionstate callback which will occur outside
of the channel_driver's monitor loop. So, on an extension state
update we have (1. global context lock ----> 2. tech_pvt lock).
Typically when we have to do a reversed locking order like this
we'd just do some sort of deadlock avoidance to fix the problem...
That will not work here. There are more locks involved here than
just the context and tech_pvt. Those are the two that are colliding,
but it is impossible to give up the context lock because the global
hints list lock MUST be held as well and we can not give that lock
up during the extensionstate callback traversal... The locking order
for the context and hints are (1. global context lock ----> 2.
hints list lock). Deadlock avoidance is not an option here.
SOLUTION:
The solution this patch implements is to queue the extension state updates
into a list and send the NOTIFY messages out during the do_monitor pvt
traversal. This clears out the problem of having to hold the context
lock before the tech_pvt lock entirely.
(closes issue #17888)
Reported by: zerohalo
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Tilghman Lesher [Fri, 10 Sep 2010 18:30:21 +0000 (18:30 +0000)]
Merged revisions 286023 via svnmerge from
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r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines
Missing newline
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Tilghman Lesher [Fri, 10 Sep 2010 05:31:31 +0000 (05:31 +0000)]
Another fix for Mac OS X.
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
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Tilghman Lesher [Fri, 10 Sep 2010 01:16:32 +0000 (01:16 +0000)]
Merged revisions 285889 via svnmerge from
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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Paul Belanger [Thu, 9 Sep 2010 22:49:19 +0000 (22:49 +0000)]
Merged revisions 285817 via svnmerge from
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r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines
GCC 4.2.x optimizations result in improper behavior of GSM codec
(closes issue #17688)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347)
Tested by: mkeuter, pprindeville
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Jason Parker [Thu, 9 Sep 2010 20:09:23 +0000 (20:09 +0000)]
Merged revisions 285742 via svnmerge from
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r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines
Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing hangups.
(closes issue #17370)
Reported by: makoto
Patches:
channel-readstring-silence-generator.patch uploaded by makoto (license 38)
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Brett Bryant [Thu, 9 Sep 2010 18:50:13 +0000 (18:50 +0000)]
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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Brett Bryant [Thu, 9 Sep 2010 17:22:25 +0000 (17:22 +0000)]
Merged revisions 285638 via svnmerge from
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r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) | 7 lines
Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class.
(closes issue #17807)
Reported by: kshumard
Review: https://reviewboard.asterisk.org/r/910/
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David Vossel [Wed, 8 Sep 2010 22:11:28 +0000 (22:11 +0000)]
Merged revisions 285566 via svnmerge from
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r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines
In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.
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David Vossel [Wed, 8 Sep 2010 21:47:29 +0000 (21:47 +0000)]
Fixes interoperability problems with session timer behavior in Asterisk.
CHANGES:
1. Never put "timer" in "Require" header. This is not to our benefit
and RFC 4028 section 7.1 even warns against it. It is possible for one
endpoint to perform session-timer refreshes while the other endpoint does
not support them. If in this case the end point performing the refreshing
puts "timer" in the Require field during a refresh, the dialog will
likely get terminated by the other end.
2. Change the behavior of 'session-timer=accept' in sip.conf (which is
the default behavior of Asterisk with no session timer configuration
specified) to only run session-timers as result of an incoming INVITE
request if the INVITE contains an "Session-Expires" header... Asterisk is
currently treating having the "timer" option in the "Supported" header as
a request for session timers by the UAC. I do not agree with this. Session
timers should only be negotiated in "accept" mode when the incoming INVITE
supplies a "Session-Expires" header, otherwise RFC 4028 says we should
treat a request containing no "Session-Expires" header as a session with
no expiration.
Below I have outlined some situations and what Asterisk's behavior is.
The table reflects the behavior changes implemented by this patch.
SITUATIONS:
-Asterisk as UAS
1. Incoming INVITE: NO "Session-Expires"
2. Incoming INVITE: HAS "Session-Expires"
-Asterisk as UAC
3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header
4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header
5. Outgoing INVITE: HAS "Session-Expires".
Active - Asterisk will have an active refresh timer regardless if the other endpoint does.
Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does.
XXXXXXX - Not possible for mode.
______________________________________
|SITUATIONS | 'session-timer' MODES |
|___________|________________________|
| | originate | accept |
|-----------|------------|-----------|
|1. | Active | Inactive |
|2. | Active | Active |
|3. | XXXXXXXX | Active |
|4. | XXXXXXXX | Inactive |
|5. | Active | XXXXXXXX |
--------------------------------------
(closes issue #17005)
Reported by: alexrecarey
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Brett Bryant [Wed, 8 Sep 2010 20:56:12 +0000 (20:56 +0000)]
Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
(closes issue #17408)
Reported by: sysreq
Patches:
asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
Tested by: sysreq
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Jason Parker [Wed, 8 Sep 2010 20:42:44 +0000 (20:42 +0000)]
Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble.
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Jason Parker [Wed, 8 Sep 2010 20:31:43 +0000 (20:31 +0000)]
Fixes issue where moh files were no longer rescanned during a reload.
(closes issue #16744)
Reported by: pj
Patches:
16744-reload.diff uploaded by qwell (license 4)
Tested by: qwell
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Tilghman Lesher [Tue, 7 Sep 2010 20:31:41 +0000 (20:31 +0000)]
Merged revisions 285365 via svnmerge from
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r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) | 9 lines
Catch invalid extensions at the parser, instead of making the core deal with them.
(closes issue #17794)
Reported by: PavelL
Patches:
20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14)
20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: PavelL
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Tilghman Lesher [Tue, 7 Sep 2010 19:07:17 +0000 (19:07 +0000)]
Merged revisions 285266 via svnmerge from
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r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) | 4 lines
Use poll, if indicated to do so, in the ast_poll2 implementation.
This fixes the unit tests on FreeBSD 8.0.
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Brett Bryant [Tue, 7 Sep 2010 17:49:07 +0000 (17:49 +0000)]
Merged revisions 285194 via svnmerge from
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r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines
Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages.
(closes issue #15726)
Reported by: 298
Patches:
M15726.diff uploaded by junky (license 177)
Tested by: junky
Review: [full review board URL with trailing slash]
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Tilghman Lesher [Mon, 6 Sep 2010 06:55:17 +0000 (06:55 +0000)]
Merged revisions 285088 via svnmerge from
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r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines
Silly convenience script for BSD platforms.
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Brett Bryant [Fri, 3 Sep 2010 18:15:49 +0000 (18:15 +0000)]
This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response.
(closes issue #17935)
Reported by: alexkuklin
Patches:
iaxshowreg uploaded by alexkuklin (license 1115)
Tested by: alexkuklin
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Terry Wilson [Fri, 3 Sep 2010 16:20:45 +0000 (16:20 +0000)]
Merged revisions 284881 via svnmerge from
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r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
code treated missing files as though they existed.
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Brett Bryant [Thu, 2 Sep 2010 20:54:33 +0000 (20:54 +0000)]
Merged revisions 284777 via svnmerge from
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r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010) | 7 lines
Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded.
(closes issue #17917)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/883/
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David Vossel [Thu, 2 Sep 2010 16:48:51 +0000 (16:48 +0000)]
Merged revisions 284703 via svnmerge from
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r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines
Removed relatedpeer code from sip_autodestruct
Handling of the relatedpeer structure associated with a
sip_pvt should be done during the final sip_destruction
function, not in sip_autodestruct.
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Tilghman Lesher [Thu, 2 Sep 2010 16:07:19 +0000 (16:07 +0000)]
Fixing build.
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Tilghman Lesher [Thu, 2 Sep 2010 05:30:16 +0000 (05:30 +0000)]
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284631
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Tilghman Lesher [Thu, 2 Sep 2010 03:57:43 +0000 (03:57 +0000)]
Failed to rerun bootstrap.sh after last commit
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Tilghman Lesher [Wed, 1 Sep 2010 22:59:50 +0000 (22:59 +0000)]
Merged revisions 284478 via svnmerge from
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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Tilghman Lesher [Wed, 1 Sep 2010 18:13:35 +0000 (18:13 +0000)]
Don't warn on floats and timestamps
(closes issue #17082)
Reported by: coolmig
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Tilghman Lesher [Tue, 31 Aug 2010 20:18:32 +0000 (20:18 +0000)]
Merged revisions 284393 via svnmerge from
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r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines
Don't send a devstate change on poke_noanswer if the state did not change.
(closes issue #17741)
Reported by: schmidts
Patches:
chan_sip.c.patch uploaded by schmidts (license 1077)
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Leif Madsen [Tue, 31 Aug 2010 18:59:31 +0000 (18:59 +0000)]
Merged revisions 284316 via svnmerge from
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r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines
Update say.conf.sample to match the rules in say.c
(closes issue #17835)
Reported by: RoadKill
Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill
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