Mark Michelson [Wed, 14 Jan 2009 20:13:48 +0000 (20:13 +0000)]
Restore the "sip show users" and "sip show user" CLI commands
(closes issue #14180)
Reported by: amorsen
Patches:
sip_show_users_161v3.diff uploaded by putnopvut (license 60)
Tested by: blitzrage, amorsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168610
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Michiel van Baak [Wed, 14 Jan 2009 19:36:57 +0000 (19:36 +0000)]
Fix compilation on FreeBSD and OSX
This started as work to fix the 'core show sysinfo'
CLI command but while working on it oej
pointed out that read_credentials did not compile neither.
So while being there, fix that as well.
Thanks for all the testing oej!
(closes issue #14129)
Reported by: ys
Tested by: oej, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168609
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Tilghman Lesher [Wed, 14 Jan 2009 19:11:14 +0000 (19:11 +0000)]
Merged revisions 168603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines
Don't read into a buffer without first checking if a value is beyond the end.
(closes issue #13600)
Reported by: atis
Patches:
20090106__bug13600.diff.txt uploaded by Corydon76 (license 14)
Tested by: atis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168604
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Tilghman Lesher [Wed, 14 Jan 2009 18:27:57 +0000 (18:27 +0000)]
Mostly spacing changes; no functionality change at all.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168601
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Mark Michelson [Wed, 14 Jan 2009 16:20:37 +0000 (16:20 +0000)]
Blocked revisions 168598 via svnmerge
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r168598 | mmichelson | 2009-01-14 10:19:26 -0600 (Wed, 14 Jan 2009) | 8 lines
Fix a logic error I found while searching through chan_agent.c
I found that the allow_multiple_logins function would never return
0 due to an incorrect comparison being used when traversing the
list of agents. While I was modifying this function, I also did
a little bit of coding guidelines cleanup, too.
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Terry Wilson [Wed, 14 Jan 2009 02:00:40 +0000 (02:00 +0000)]
Merged revisions 168593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128.
Proper bounds checking was not in place to enforce this limit, so if more than
128 extensions were passed to the Page() app, Asterisk would crash. This patch
instead dynamically allocates memory for the ast_dial structures and removes
the (non-functional) arbitrary limit.
This issue would have special importance to anyone who is dynamically creating
the argument passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external interface.
The patch posted by a_villacis was slightly modified for some coding guidelines
and other cleanups. Thanks, a_villacis!
(closes issue #14217)
Reported by: a_villacis
Patches:
20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
Tested by: otherwiseguy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594
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Tilghman Lesher [Tue, 13 Jan 2009 23:57:46 +0000 (23:57 +0000)]
Janitor patch for chan_misdn (make channel variable access safe)
(closes issue #12887)
Reported by: pputman
Patches:
chan_misdn_threadsafe.patch uploaded by pputman (license 81)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168591
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Terry Wilson [Tue, 13 Jan 2009 23:05:43 +0000 (23:05 +0000)]
Fully overwrite a same-named file when uploading
(closes issue #14190)
Reported by: timking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168588
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Terry Wilson [Tue, 13 Jan 2009 23:00:27 +0000 (23:00 +0000)]
Add option to hide console connect messages
(closes issue #14222)
Reported by: jamesgolovich
Patches:
asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176)
Tested by: otherwiseguy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168585
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Mark Michelson [Tue, 13 Jan 2009 22:30:59 +0000 (22:30 +0000)]
Clarify a message that app_queue prints and change to a debug-level message
The "No one is answering..." verbose message contained 3 numbers that were not
explained in any way to whoever was viewing the message. It is more helpful now
since the message explains what the numbers mean. Also, the message has been
downgraded to "DEBUG" level.
(closes issue #14172)
Reported by: caio1982
Patches:
queue_answering_debug.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168579
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Terry Wilson [Tue, 13 Jan 2009 22:22:34 +0000 (22:22 +0000)]
Merged revisions 168551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines
Don't pass a value with a side effect to a macro
(closes issue #14176)
Reported by: paraeco
Patches:
chan_sip.c.diff uploaded by paraeco (license 658)
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Mark Michelson [Tue, 13 Jan 2009 21:18:13 +0000 (21:18 +0000)]
Allow specifying a port number in the user portion of a register => line in sip.conf
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575
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Russell Bryant [Tue, 13 Jan 2009 19:22:13 +0000 (19:22 +0000)]
Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines
Revert unnecessary indications API change from rev 122314
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Tilghman Lesher [Tue, 13 Jan 2009 17:51:12 +0000 (17:51 +0000)]
Merged revisions 168546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) | 6 lines
If either conditional is NULL, don't try copying it.
(closes issue #14226)
Reported by: caspy
Patches:
20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168547
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Dwayne M. Hubbard [Tue, 13 Jan 2009 16:02:13 +0000 (16:02 +0000)]
correct a CLI description
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168539
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Tilghman Lesher [Mon, 12 Jan 2009 23:45:51 +0000 (23:45 +0000)]
Merged revisions 167095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines
Repeat attempts to write when we receive -EAGAIN from the driver, as detailed
in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
Reported by: Jerry Geis (via the -users list)
Fixed by: me (license 14)
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Mark Michelson [Mon, 12 Jan 2009 23:12:30 +0000 (23:12 +0000)]
bump the verbosity of a message in srv.c up by one. It used to be
at this level prior to a large patch merge which converted ast_verbose
calls to ast_verb
(closes issue #14221)
Reported by: jcovert
Patches:
srv.c.patch uploaded by jcovert (license 551)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168523
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Tilghman Lesher [Mon, 12 Jan 2009 23:06:12 +0000 (23:06 +0000)]
Some platforms (notably, the BSDs) have a more efficient implementation called
closefrom(3).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168522
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Jeff Peeler [Mon, 12 Jan 2009 21:51:46 +0000 (21:51 +0000)]
Merged revisions 168516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines
(closes issue #13881)
Reported by: hoowa
Update the app CDR field for AGI commands that are not executing an application via "exec".
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Jeff Peeler [Mon, 12 Jan 2009 20:53:04 +0000 (20:53 +0000)]
Merged revisions 168507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines
(closes issue #12269)
Reported by: IgorG
Tested by: denisgalvao
This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock.
Review: http://reviewboard.digium.com/r/35/
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Olle Johansson [Mon, 12 Jan 2009 16:31:27 +0000 (16:31 +0000)]
Better to use the proper app name
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168497
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Mark Michelson [Mon, 12 Jan 2009 15:00:00 +0000 (15:00 +0000)]
Merged revisions 168482 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan 2009) | 5 lines
I am reverting the fix made in revision 168128 (and its upward merges)
after being contacted by Olle Johansson and being shown how this fix is
incorrect. Thanks to Olle for clearing this up for me.
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Russell Bryant [Mon, 12 Jan 2009 14:57:49 +0000 (14:57 +0000)]
Merged revisions 168480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines
s/ringdance/ringcadence/ for Bulgaria
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Olle Johansson [Mon, 12 Jan 2009 14:35:09 +0000 (14:35 +0000)]
Don't include swap.h unless we have swapctl
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168479
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Tilghman Lesher [Sat, 10 Jan 2009 01:42:45 +0000 (01:42 +0000)]
sizeof for a stringfield is 4. Kinda low for reconstructing a field value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168334
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Kevin P. Fleming [Fri, 9 Jan 2009 23:16:08 +0000 (23:16 +0000)]
Merged revisions 168267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line
update to use new sound file packages that include license files
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Richard Mudgett [Fri, 9 Jan 2009 23:15:26 +0000 (23:15 +0000)]
Spacing change
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168269
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Michiel van Baak [Fri, 9 Jan 2009 23:04:46 +0000 (23:04 +0000)]
Add a script to find out the correct settings for Asterisk behind NAT
(closes issue #13065)
Reported by: tzafrir
Patches:
sip_nat_settings uploaded by tzafrir (license 46)
sip_nat_settings_6 uploaded by mvanbaak (license 7)
Tested by: tzafrir, pabelanger, Dovid and moi
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168265
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Russell Bryant [Fri, 9 Jan 2009 22:21:05 +0000 (22:21 +0000)]
Merged revisions 168198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines
Make this compile for mvanbaak
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Mark Michelson [Fri, 9 Jan 2009 21:53:26 +0000 (21:53 +0000)]
Merged revisions 168128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines
Add check_via calls to more request handlers
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.
(closes issue #13071)
Reported by: baron
Patches:
check_via.patch uploaded by baron (license 531)
Tested by: baron
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Richard Mudgett [Fri, 9 Jan 2009 21:43:30 +0000 (21:43 +0000)]
Merged revisions 168191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines
* Fix for JIRA AST-175/ABE-1757
* Miscellaneous doxygen comments added.
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Terry Wilson [Fri, 9 Jan 2009 20:25:25 +0000 (20:25 +0000)]
Don't leak memory if phoneprov.conf does not exist
(closes issue #14203)
Reported by: jamesgolovich
Patches:
asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich (license 176)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168142
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Tilghman Lesher [Fri, 9 Jan 2009 18:30:55 +0000 (18:30 +0000)]
When using ast_str with a non-ast_str-enabled API, we need to update the buffer
or otherwise, we cannot use ast_str_strlen().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168090
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Matthew Nicholson [Fri, 9 Jan 2009 18:01:26 +0000 (18:01 +0000)]
Added a comment to logger.c about where to put includes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168054
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Matthew Nicholson [Fri, 9 Jan 2009 16:32:34 +0000 (16:32 +0000)]
Use ast_safe_system() in logger.c instead of system()
(closes issue #14194)
Reported by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168014
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Terry Wilson [Fri, 9 Jan 2009 01:15:43 +0000 (01:15 +0000)]
Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167973
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Terry Wilson [Fri, 9 Jan 2009 00:13:12 +0000 (00:13 +0000)]
Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167935
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Tilghman Lesher [Thu, 8 Jan 2009 22:37:20 +0000 (22:37 +0000)]
Merged revisions 167840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines
Don't truncate database results at 255 chars.
(closes issue #14069)
Reported by: evandro
Patches:
20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
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Mark Michelson [Thu, 8 Jan 2009 22:34:52 +0000 (22:34 +0000)]
Revert chan_sip changes which were accidentally committed
in revision 167792
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167888
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Tilghman Lesher [Thu, 8 Jan 2009 21:40:28 +0000 (21:40 +0000)]
Fix variables to comply with documentation changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167837
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Tilghman Lesher [Thu, 8 Jan 2009 21:32:45 +0000 (21:32 +0000)]
Textual changes, consistency in status variable naming, and other minor bugs.
(closes issue #13943)
Reported by: Marquis
Patches:
minivm_trunk_fixes3.patch uploaded by Marquis (license 32)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167835
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Mark Michelson [Thu, 8 Jan 2009 19:48:42 +0000 (19:48 +0000)]
Add the average talk time for a queue
This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.
(closes issue #13960)
Reported by: coolmig
Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167792
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Tilghman Lesher [Thu, 8 Jan 2009 19:44:19 +0000 (19:44 +0000)]
Convert dialplan application DAHDISendCallreroutingFacility to use commas.
(closes issue #13836)
Reported by: eliel
Patches:
chan_dahdi.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167791
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Kevin P. Fleming [Thu, 8 Jan 2009 17:26:03 +0000 (17:26 +0000)]
Merged revisions 167714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line
remove an unnecessary argument to queue_request()
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Kevin P. Fleming [Thu, 8 Jan 2009 16:43:26 +0000 (16:43 +0000)]
Merged revisions 167620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines
When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available.
http://reviewboard.digium.com/r/123/
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Leif Madsen [Thu, 8 Jan 2009 14:27:53 +0000 (14:27 +0000)]
Oops... fix the fieldname I changed yesterday to be right.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167662
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Russell Bryant [Wed, 7 Jan 2009 22:36:34 +0000 (22:36 +0000)]
Merged revisions 167566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines
Fix the last couple of places where free() was improperly used directly.
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Russell Bryant [Wed, 7 Jan 2009 22:27:23 +0000 (22:27 +0000)]
Merged revisions 167554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines
Don't fclose() the file early, the filestream destructor will handle it.
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Russell Bryant [Wed, 7 Jan 2009 22:20:31 +0000 (22:20 +0000)]
Merged revisions 167545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines
Only try to close the file if one was actually opened
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167546
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Russell Bryant [Wed, 7 Jan 2009 22:05:29 +0000 (22:05 +0000)]
Merged revisions 167541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines
Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object.
Reported by JunK-Y on IRC, #asterisk-dev
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167542
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BJ Weschke [Wed, 7 Jan 2009 18:20:31 +0000 (18:20 +0000)]
Answer the channel if it has not already been answered and we've already found a valid profile for followme.
(closes issue #14140)
Reported by: dimas
Patches:
14140.patch uploaded by dimas
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167478
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Leif Madsen [Wed, 7 Jan 2009 18:18:45 +0000 (18:18 +0000)]
Update queues.conf.sample documentation.
Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
(closes issue #14179)
Reported by: CrashHD
Tested by: CrashHD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167477
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Russell Bryant [Wed, 7 Jan 2009 17:35:39 +0000 (17:35 +0000)]
Merged revisions 167432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) | 4 lines
Treat an empty string the same way as a NULL country argument.
In passing, simplify the handling of returning a default tone zone.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167442
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Doug Bailey [Wed, 7 Jan 2009 17:05:37 +0000 (17:05 +0000)]
Cleanup fsk spill if off hook is detected during mwi spill.
Correct logic error in handling events when sending mwi spill
(closes issue #14143)
Reported by: alecdavis
Patches:
chan_dahdi.handle_init_event2.diff.txt uploaded by dbailey
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167416
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Leif Madsen [Wed, 7 Jan 2009 14:26:19 +0000 (14:26 +0000)]
Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167373
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Mark Michelson [Tue, 6 Jan 2009 21:36:44 +0000 (21:36 +0000)]
Merged revisions 167299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) | 8 lines
Use the correct variable when creating the format string
(closes issue #14177)
Reported by: nic_bellamy
Patches:
asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167301
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Tilghman Lesher [Tue, 6 Jan 2009 21:02:33 +0000 (21:02 +0000)]
Merged revisions 167260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167260 | tilghman | 2009-01-06 14:48:05 -0600 (Tue, 06 Jan 2009) | 9 lines
Merged revisions 167259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines
Security fix AST-2009-001.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167265
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Mark Michelson [Mon, 5 Jan 2009 16:59:36 +0000 (16:59 +0000)]
Merged revisions 167179 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines
A couple of changes to T.38 SDP attribute handling
There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.
It was also discovered that a particular type of
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.
Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.
(closes issue #13976)
Reported by: linulin
Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167180
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Tilghman Lesher [Mon, 5 Jan 2009 16:44:47 +0000 (16:44 +0000)]
More clearly explain that quote marks are no longer necessary.
(closes issue #13718)
Reported by: davidw
Patches:
20081020__bug13718.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167176
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Jeff Peeler [Sat, 3 Jan 2009 20:29:54 +0000 (20:29 +0000)]
When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167125
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Sean Bright [Wed, 31 Dec 2008 23:07:14 +0000 (23:07 +0000)]
Mostly just whitespace, but also convert 'CVS' to 'SVN' in a couple
places and fix a few typos I found in the CODING_GUIDELINES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167061
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Terry Wilson [Wed, 31 Dec 2008 22:53:55 +0000 (22:53 +0000)]
Don't forget to free typename
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167057
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Mark Michelson [Wed, 31 Dec 2008 21:52:02 +0000 (21:52 +0000)]
Change some incorrect syntax for pri set debug and correct
an off-by-one error in ss7 set debug command
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167021
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Tilghman Lesher [Wed, 31 Dec 2008 19:39:05 +0000 (19:39 +0000)]
That was weird...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166958
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Tilghman Lesher [Wed, 31 Dec 2008 19:34:28 +0000 (19:34 +0000)]
Merged revisions 166953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines
Also inherit the musiconhold class.
(Closes #14153)
Reported by: Jerry Geis, via the users list.
Patch by: me (license 14)
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Terry Wilson [Tue, 30 Dec 2008 20:50:05 +0000 (20:50 +0000)]
Fix some svn:keywords
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166908
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Mark Michelson [Mon, 29 Dec 2008 18:04:52 +0000 (18:04 +0000)]
Update app_queue to deal with the removal of AST_PBX_KEEPALIVE
When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.
I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166861
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Eliel C. Sardanons [Sun, 28 Dec 2008 15:36:25 +0000 (15:36 +0000)]
Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166823
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Russell Bryant [Sun, 28 Dec 2008 15:15:14 +0000 (15:15 +0000)]
Merged revisions 166772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines
Use strncat() instead of an sprintf() in which source and target buffers overlap
http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
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Terry Wilson [Wed, 24 Dec 2008 15:10:42 +0000 (15:10 +0000)]
There is no section 22.2.2 in rfc 3261. I believe 26.2.2 is what was meant:
Note that in the SIPS URI scheme, transport is independent of TLS,
and thus "sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166731
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Tilghman Lesher [Tue, 23 Dec 2008 20:47:08 +0000 (20:47 +0000)]
Allow semicolons and extended characters in user-specified SIP headers.
(closes issue #14110)
Reported by: gork
Patches:
20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: gork, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166696
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Steve Murphy [Tue, 23 Dec 2008 18:13:49 +0000 (18:13 +0000)]
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
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r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
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Russell Bryant [Tue, 23 Dec 2008 16:04:54 +0000 (16:04 +0000)]
Fix spelling error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166625
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Tilghman Lesher [Tue, 23 Dec 2008 15:41:42 +0000 (15:41 +0000)]
Blocked revisions 166592 via svnmerge
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r166592 | tilghman | 2008-12-23 09:35:38 -0600 (Tue, 23 Dec 2008) | 3 lines
Compile, even if both DAHDI and Zaptel are not installed.
(Closes issue #14120)
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Mark Michelson [Tue, 23 Dec 2008 15:17:54 +0000 (15:17 +0000)]
Merged revisions 166568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines
Fix a crash resulting from a datastore with inheritance but no duplicate callback
The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.
(closes issue #14113)
Reported by: francesco_r
Patches:
14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166569
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Tilghman Lesher [Tue, 23 Dec 2008 04:32:15 +0000 (04:32 +0000)]
Merged revisions 166509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines
Use the integer form of condition for integer comparisons.
(closes issue #14127)
Reported by: andrew
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166533
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Mark Michelson [Mon, 22 Dec 2008 23:25:34 +0000 (23:25 +0000)]
Always use the value of the AGISIGHUP when running an AGI.
Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel.
(closes issue #13711)
Reported by: fmueller
Patches:
13711.patch uploaded by putnopvut (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166470
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Russell Bryant [Mon, 22 Dec 2008 21:45:28 +0000 (21:45 +0000)]
Cosmetic change - don't mix struct initializer styles.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166436
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Mark Michelson [Mon, 22 Dec 2008 21:08:03 +0000 (21:08 +0000)]
Merged revisions 166380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines
Fix a deadlock relating to channel locks and autoservice
It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of
autoservice.
The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.
The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.
In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.
It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.
(closes issue #14057)
Reported by: rtrauntvein
Patches:
14057v3.patch uploaded by putnopvut (license 60)
Tested by: rtrauntvein
Review: http://reviewboard.digium.com/r/107/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166382
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Russell Bryant [Mon, 22 Dec 2008 20:26:48 +0000 (20:26 +0000)]
Fix a bad typo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166377
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Russell Bryant [Mon, 22 Dec 2008 17:44:23 +0000 (17:44 +0000)]
Remove some error messages. This is the default handler that is valid to use.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166342
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Russell Bryant [Mon, 22 Dec 2008 17:29:10 +0000 (17:29 +0000)]
Merged revisions 166297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines
Fix up timeout handling in ast_carefulwrite().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166317
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Russell Bryant [Mon, 22 Dec 2008 17:09:36 +0000 (17:09 +0000)]
Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.
This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes. It is currently used in AMI to
address the issue that has been reported. However, there are probably a few
other places where this could be used.
(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166282
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Russell Bryant [Mon, 22 Dec 2008 16:10:40 +0000 (16:10 +0000)]
Re-work ref count handling of MoH classes using astobj2 to resolve crashes.
(closes issue #13566)
Reported by: igorcarneiro
Tested by: russell
Review: http://reviewboard.digium.com/r/106/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166273
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Joshua Colp [Mon, 22 Dec 2008 16:08:13 +0000 (16:08 +0000)]
Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port.
(closes issue #13628)
Reported by: pananix
Patches:
bug13628.patch uploaded by jpeeler (license 325)
Tested by: file, blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166268
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Mark Michelson [Mon, 22 Dec 2008 16:07:59 +0000 (16:07 +0000)]
Fix a file playback crash and explicitly initialize values in func_timeout.c
A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.
A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.
(closes issue #14118)
Reported by: blitzrage
Patches:
14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166267
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Russell Bryant [Mon, 22 Dec 2008 14:52:35 +0000 (14:52 +0000)]
Blocked revisions 166262 via svnmerge
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r166262 | russell | 2008-12-22 08:45:27 -0600 (Mon, 22 Dec 2008) | 7 lines
Re-work ref count handling of MoH classes using astobj2 to resolve crashes.
(closes issue #13566)
Reported by: igorcarneiro
Tested by: russell
Review: http://reviewboard.digium.com/r/106/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166263
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Russell Bryant [Mon, 22 Dec 2008 14:16:54 +0000 (14:16 +0000)]
Remove AST_PBX_KEEPALIVE usage from res_agi.
This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage
was for the AGI command, "asyncagi break". This patch removes this feature.
Normally, a feature would not be removed like this. However, this code is
broken and usage of it will result in a memory leak.
Usage of this feature will make the AGI code return a result of
AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed
ownership of the channel. The channel thread will exit without destroying the
channel. Unfortunately, _no_ thread has ownership of the channel at this
point. There are a couple of serious problems here:
1) The only way to recover the caller is to issue a channel redirect. This
will work, but this will be done with a masquerade, and the old ast_channel
structure will be lost.
2) Until the channel redirect happens, there is no code servicing the channel.
That means nothing is reading audio or handling events coming from the
channel. This is very bad.
The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code. That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166258
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Russell Bryant [Sat, 20 Dec 2008 01:37:23 +0000 (01:37 +0000)]
Make a note about formatting the review URL in commit messages
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166219
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Mark Michelson [Fri, 19 Dec 2008 23:45:00 +0000 (23:45 +0000)]
Get rid of an extra space.
I don't know how this crept back in when I had already
fixed it earlier
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166162
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Mark Michelson [Fri, 19 Dec 2008 23:36:22 +0000 (23:36 +0000)]
Blocked revisions 166157 via svnmerge
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r166157 | mmichelson | 2008-12-19 17:34:57 -0600 (Fri, 19 Dec 2008) | 9 lines
Backport of AUDIOHOOK_INHERIT for Asterisk 1.4
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166158
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Mark Michelson [Fri, 19 Dec 2008 22:40:57 +0000 (22:40 +0000)]
Remove the verbatim tag from the author line
I could have sworn I already did that before, though...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166095
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Mark Michelson [Fri, 19 Dec 2008 22:26:16 +0000 (22:26 +0000)]
Adding a new dialplan function AUDIOHOOK_INHERIT
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/102/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092
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Matthew Fredrickson [Fri, 19 Dec 2008 21:44:18 +0000 (21:44 +0000)]
Add configuration support for half_full DAHDI buffer policy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166058
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Jeff Peeler [Fri, 19 Dec 2008 19:49:25 +0000 (19:49 +0000)]
Blocked revisions 165991 via svnmerge
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r165991 | jpeeler | 2008-12-19 13:48:00 -0600 (Fri, 19 Dec 2008) | 15 lines
(closes issue #13480)
Reported by: tzafrir
Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug:
#ifdef HAVE_ZAPTEL
fd = open("/dev/zap/pseudo", O_RDWR);
#else
fd = open("/dev/dahdi/pseudo", O_RDWR);
#endif
is replaced with:
fd = open(DAHDI_FILE_PSEUDO, O_RDRW);
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165992
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Eliel C. Sardanons [Fri, 19 Dec 2008 18:20:46 +0000 (18:20 +0000)]
Fix the XML documentation for Record().
<value> tags inside <variable> elements must have CDATA and no
another XML node.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165954
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Russell Bryant [Fri, 19 Dec 2008 15:05:09 +0000 (15:05 +0000)]
Merged revisions 165889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines
Ensure that the chanspy datastore is fully initialized.
This patch resolved some random crash issues observed by a user on a BSD system
(closes issue #14111)
Reported by: ys
Patches:
app_chanspy.c.diff uploaded by ys (license 281)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165890
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Russell Bryant [Fri, 19 Dec 2008 14:48:35 +0000 (14:48 +0000)]
Disable some automatic links generated by doxygen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165886
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Russell Bryant [Fri, 19 Dec 2008 14:42:51 +0000 (14:42 +0000)]
Introduce commit message formatting guidelines.
This documents the recommended outline to use for commit message. It also
covers information on special tags that can be used in commit messages, as well
as the template functionality that is available on bugs.digium.com.
Review: http://reviewboard.digium.com/r/96/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165883
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Russell Bryant [Thu, 18 Dec 2008 21:44:47 +0000 (21:44 +0000)]
Merged revisions 165796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines
Make ast_carefulwrite() be more careful.
This patch handles some additional cases that could result in partial writes
to the file description. This was done to address complaints about partial
writes on AMI.
(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165801
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