Paul Belanger [Wed, 7 Sep 2011 19:38:58 +0000 (19:38 +0000)]
Merged revisions 334844 via svnmerge from
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r334844 | pabelanger | 2011-09-07 15:37:24 -0400 (Wed, 07 Sep 2011) | 11 lines
Merged revisions 334843 via svnmerge from
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r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines
Cleanup chan_iax2.c log messages
Review: https://code.asterisk.org/code/cru/CR-AST-11
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Richard Mudgett [Wed, 7 Sep 2011 19:35:18 +0000 (19:35 +0000)]
Merged revisions 334841 via svnmerge from
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r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines
Merged revisions 334840 via svnmerge from
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r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines
Fix AMI action Park crash.
* Made AMI action Park not say anything to the parker channel (AMI header
Channel2) since the AMI action is a third party parking the call. (This
is a change in behavior that cannot be preserved without a lot of effort.)
* Made not play pbx-parkingfailed if the Park 's' option is used.
JIRA AST-660
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Stefan Schmidt [Wed, 7 Sep 2011 15:37:32 +0000 (15:37 +0000)]
Merged revisions 334747 via svnmerge from
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r334747 | schmidts | 2011-09-07 15:10:37 +0000 (Wed, 07 Sep 2011) | 9 lines
Merged revisions 334682 via svnmerge from
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r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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Stefan Schmidt [Wed, 7 Sep 2011 14:47:03 +0000 (14:47 +0000)]
clean up wrong merged stuff
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334744
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Stefan Schmidt [Wed, 7 Sep 2011 14:23:38 +0000 (14:23 +0000)]
Merged revisions 334682 via svnmerge from
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r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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Stefan Schmidt [Wed, 7 Sep 2011 13:31:13 +0000 (13:31 +0000)]
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334683
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Alec L Davis [Wed, 7 Sep 2011 08:17:24 +0000 (08:17 +0000)]
Merged revisions 334621 via svnmerge from
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r334621 | alecdavis | 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) | 9 lines
Merged revisions 334620 via svnmerge from
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r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines
peroid typo
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Alec L Davis [Wed, 7 Sep 2011 08:06:32 +0000 (08:06 +0000)]
log Asterisk Version number, Build etc into each log file
Allow tracking of previous versions through log file records to be tracked.
Each time log file is created or opened, log Asterisk Version, Buildinfo. etc.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1409/
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Alec L Davis [Wed, 7 Sep 2011 07:48:25 +0000 (07:48 +0000)]
Merged revisions 334617 via svnmerge from
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r334617 | alecdavis | 2011-09-07 19:45:00 +1200 (Wed, 07 Sep 2011) | 17 lines
Merged revisions 334616 via svnmerge from
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r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines
Prevent segfault if call arrives before Asterisk is fully booted.
Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk
is fully booted.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1407/
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Tilghman Lesher [Wed, 7 Sep 2011 00:54:36 +0000 (00:54 +0000)]
Implement the '!' negation element to negate codecs directly in the allow keyword.
This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574
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Gregory Nietsky [Tue, 6 Sep 2011 16:15:50 +0000 (16:15 +0000)]
Merged revisions 334455 via svnmerge from
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r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines
Merged revisions 334453 via svnmerge from
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r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines
Make SQL query in app_voicemail.c portable LIMIT is not portable.
Regression from r312212
(closes issue ASTERISK-18255)
Reported by: Leif Madsen
Tested by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1415/
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Paul Belanger [Tue, 6 Sep 2011 16:08:10 +0000 (16:08 +0000)]
Merged revisions 334514 via svnmerge from
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r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep 2011) | 6 lines
authdebug is now disabled by default
To enable this functionaility again set authdebug = yes in iax.conf
Review: https://reviewboard.asterisk.org/r/1414/
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Gregory Nietsky [Tue, 6 Sep 2011 16:04:02 +0000 (16:04 +0000)]
Revert r334472 due to properties going missing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334515
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Gregory Nietsky [Tue, 6 Sep 2011 14:24:07 +0000 (14:24 +0000)]
Merged revisions 334455 via svnmerge from
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r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines
Merged revisions 334453 via svnmerge from
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r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines
Make SQL query in app_voicemail.c portable LIMIT is not portable.
Regression from r312212
(closes issue ASTERISK-18255)
Reported by: Leif Madsen
Tested by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1415/
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Richard Mudgett [Fri, 2 Sep 2011 21:09:31 +0000 (21:09 +0000)]
Merged revisions 334357 via svnmerge from
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r334357 | rmudgett | 2011-09-02 16:08:16 -0500 (Fri, 02 Sep 2011) | 26 lines
Merged revisions 334355 via svnmerge from
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r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines
MusicOnHold has extra unref which may lead to memory corruption and crash.
The problem happens when a call is disconnected and you had started a MOH
class that does not use the files mode. If you define REF_DEBUG and
recreate the problem, it will announce itself with the following warning:
Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained,
and class is still in a container!
* Fixed moh_alloc() and moh_release() functions not handling the
state->class reference consistently.
(closes issue ASTERISK-18346)
Reported by: Mark Murawski
Patches:
jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Mark Murawski
Review: https://reviewboard.asterisk.org/r/1404/
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Richard Mudgett [Fri, 2 Sep 2011 17:19:17 +0000 (17:19 +0000)]
Merged revisions 334297 via svnmerge from
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r334297 | rmudgett | 2011-09-02 12:15:08 -0500 (Fri, 02 Sep 2011) | 46 lines
Merged revisions 334296 via svnmerge from
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r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines
Fix potential memory allocation failure crashes in config.c.
* Added required checks to the returned memory allocation pointers to
prevent crashes.
* Made ast_include_rename() create a replacement ast_variable list node if
the new filename is longer than the available space. Fixes potential
crash and memory leak.
* Factored out ast_variable_move() from ast_variable_update() so
ast_include_rename() can also use it when creating a replacement
ast_variable list node.
* Made the filename stuffed at the end of the struct a minimum allocated
size in ast_variable_new() in case ast_include_rename() changes the stored
filename.
* Constify struct char pointers pointing to strings stuffed at the end of
the struct for: ast_variable, cache_file_mtime, and ast_config_map.
* Factored out cfmtime_new() to remove inlined code and allow some struct
pointers to become const.
* Removed the list lock from struct cache_file_mtime that was never used.
* Added doxygen comments to several structure elements and better
documented what strings are stuffed at the struct end char array.
* Reworked ast_config_text_file_save() and set_fn() to handle allocation
failure of the include file scratch pad object tracking blank lines.
* Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure
it is long enough for any filename with path. Also reduced the number of
container fileset buckets from a rediculus 180,000 to 1023.
JIRA AST-618
Review: https://reviewboard.asterisk.org/r/1378/
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Tilghman Lesher [Thu, 1 Sep 2011 17:41:09 +0000 (17:41 +0000)]
Merged revisions 334235 via svnmerge from
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r334235 | tilghman | 2011-09-01 12:39:32 -0500 (Thu, 01 Sep 2011) | 9 lines
Merged revisions 334234 via svnmerge from
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r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011) | 2 lines
Remove 1.6 compatibility documentation from 1.8, as it no longer applies.
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Tilghman Lesher [Thu, 1 Sep 2011 17:31:34 +0000 (17:31 +0000)]
Merged revisions 334230 via svnmerge from
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r334230 | tilghman | 2011-09-01 12:30:19 -0500 (Thu, 01 Sep 2011) | 25 lines
Merged revisions 334229 via svnmerge from
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r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines
Create a local alias for ast_odbc_clear_cache.
As a function pointer, the reference has to be resolved at load time
irrespective of the RTLD_LAZY flag. Creating a local alias solves
this problem, because the structure is initialized with that local
function pointer, while the actual function can remain lazily linked
until runtime.
The reason why this is important is because we lazily load function
references during the module loading process, in order to obtain
priority values for each module, ensuring that modules are loaded in
the correct order. Previous to this change, when this module was
initially loaded, the module loader would emit a symbol resolution
error, because of the above requirement.
Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
Walter Doekes, patch by me)
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Matthew Nicholson [Wed, 31 Aug 2011 18:54:33 +0000 (18:54 +0000)]
Merged revisions 334157 via svnmerge from
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r334157 | mnicholson | 2011-08-31 13:53:40 -0500 (Wed, 31 Aug 2011) | 11 lines
Merged revisions 334156 via svnmerge from
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r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines
Disable T.38 when we get a invite with image media port set to 0
ASTERISK-17678
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Richard Mudgett [Wed, 31 Aug 2011 18:11:23 +0000 (18:11 +0000)]
Optimize chan_sip.c check_rtp_timeout() function.
* Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.
(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1377/
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Matthew Nicholson [Wed, 31 Aug 2011 16:31:30 +0000 (16:31 +0000)]
Merged revisions 334064 via svnmerge from
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r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug 2011) | 4 lines
only alter the gateway_timeout when attching the gateway to a channel
ASTERISK-18219
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Richard Mudgett [Wed, 31 Aug 2011 16:02:11 +0000 (16:02 +0000)]
Merged revisions 334013 via svnmerge from
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r334013 | rmudgett | 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) | 30 lines
Merged revisions 334012 via svnmerge from
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r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines
No DAHDI channel available for conference, user introduction disabled.
The following error will consistently occur when trying to dial into a
MeetMe conference when the server does not have DAHDI hardware installed:
app_meetme.c: No DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?)
While chan_dahdi is loaded correctly during compilation and install of
Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if hardware
does not exist, causing MeetMe to be unable to open a DAHDI pseudo
channel.
* Allow chan_dahdi to create a pseudo channel when there is no
chan_dahdi.conf file to load.
(closes issue ASTERISK-17398)
Reported by: Preston Edwards
Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
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Richard Mudgett [Wed, 31 Aug 2011 15:25:35 +0000 (15:25 +0000)]
Merged revisions 334010 via svnmerge from
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r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines
Merged revisions 334009 via svnmerge from
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r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines
Call pickup race leaves orphaned channels or crashes.
Multiple users attempting to pickup a call that has been forked to
multiple extensions either crashes or fails a masquerade with a "bad
things may happen" message.
This is the scenario that is causing all the grief:
1) Pickup target is selected
2) target is marked as being picked up in ast_do_pickup()
3) target is unlocked by ast_do_pickup()
4) app dial or queue gets a chance to hang up losing calls and calls
ast_hangup() on target
5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
ast_channel_masquerade(), ast_hangup() completes successfully and the
channel is no longer in the channels container.
6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the
masquerade on the dead channel.
7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel
8) bad things happen while doing the masquerade and in the process
ast_do_masquerade() puts the dead channel back into the channels container
9) The "orphaned" channel is visible in the channels list if a crash does
not happen.
This patch does the following:
* Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel
and not release the channel lock until that has happened.
* Made __ast_channel_masquerade() not setup a masquerade if either channel
has AST_FLAG_ZOMBIE set.
* Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work.
(closes issue ASTERISK-18222)
Reported by: Alec Davis
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
(closes issue ASTERISK-18273)
Reported by: Karsten Wemheuer
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
Review: https://reviewboard.asterisk.org/r/1400/
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Kinsey Moore [Wed, 31 Aug 2011 15:20:21 +0000 (15:20 +0000)]
Merged revisions 334007 via svnmerge from
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r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines
Merged revisions 334006 via svnmerge from
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r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines
Correct an AMI protocol violation with SIPshowpeer
The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
ended by using \r\n this confuses any interfacing script.
(closes issue ASTERISK-17486)
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Alexandr Anikin [Tue, 30 Aug 2011 22:16:13 +0000 (22:16 +0000)]
Merged revisions 333961-333962 via svnmerge from
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r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines
Merged revisions 333947 via svnmerge from
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r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines
cleanups in ACF/ARJ GK replies processing
fixed long (24 sec) pause if acf/arj proccessed
before ast_cond_wait called to wait this
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r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines
security fix. really drop call if signalling addr is not same as socket
addr
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Matthew Nicholson [Tue, 30 Aug 2011 14:03:02 +0000 (14:03 +0000)]
Merged revisions 333895 via svnmerge from
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r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug 2011) | 6 lines
Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway).
Patch by: irroot
Review: https://reviewboard.asterisk.org/r/1385/
ASTERISK-18219
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Terry Wilson [Mon, 29 Aug 2011 21:43:33 +0000 (21:43 +0000)]
Merged revisions 333837 via svnmerge from
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r333837 | twilson | 2011-08-29 16:41:13 -0500 (Mon, 29 Aug 2011) | 22 lines
Merged revisions 333836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines
Refresh peer address if DNS unavailable at peer creation
If Asterisk starts and no DNS is available, outbound registrations will fail
indefinitely. This patch copies the address from the sip_registry struct, which
will be updated, to the peer->addr when necessary.
If dnsmgr is enabled, the registration fails without the patch because even
though the address on the registry is updated via dnsmgr, the address is just
copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update
the address that is copied to the sip_pvt or peers.
Closes issue ASTERISK-18000
Review: https://reviewboard.asterisk.org/r/1335/
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Richard Mudgett [Mon, 29 Aug 2011 21:17:51 +0000 (21:17 +0000)]
Merged revisions 333786 via svnmerge from
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r333786 | rmudgett | 2011-08-29 16:12:29 -0500 (Mon, 29 Aug 2011) | 13 lines
Merged revisions 333784-333785 via svnmerge from
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r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines
Fix deadlock potential of chan_mobile.c:mbl_ast_hangup().
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r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line
Add some do not hold locks notes to channel.h
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Matthew Nicholson [Mon, 29 Aug 2011 18:28:02 +0000 (18:28 +0000)]
Merged revisions 333716 via svnmerge from
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r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug 2011) | 5 lines
It is possible for the gateway to be attached when the channel is still
negotiating T.38. This change handles that case.
ASTERISK-18329
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Terry Wilson [Mon, 29 Aug 2011 17:31:40 +0000 (17:31 +0000)]
Merged revisions 333681 via svnmerge from
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r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011) | 7 lines
Use realtime text when it is negotiated
This patch make use of wirte_text() realtime text instead of
send_text() if T.140 is in native formats. ASTERISK-17937
Review: https://reviewboard.asterisk.org/r/1356/
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Matthew Jordan [Mon, 29 Aug 2011 17:14:26 +0000 (17:14 +0000)]
Merged revisions 333631 via svnmerge from
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r333631 | mjordan | 2011-08-29 12:12:55 -0500 (Mon, 29 Aug 2011) | 9 lines
Merged revisions 333630 via svnmerge from
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r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) | 1 line
Fixed improperly formatted TestEvent AMI message in app_voicemail
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Jonathan Rose [Mon, 29 Aug 2011 15:58:24 +0000 (15:58 +0000)]
Merged revisions 333570 via svnmerge from
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r333570 | jrose | 2011-08-29 10:56:56 -0500 (Mon, 29 Aug 2011) | 11 lines
Merged revisions 333569 via svnmerge from
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r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | 4 lines
Accidental use of variable client->status instead of client->state in from ASTERISK-18078
(issue ASTERISK-18078)
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Tzafrir Cohen [Sun, 28 Aug 2011 09:57:47 +0000 (09:57 +0000)]
chan_vpb: remove unused variables (gcc4.6)
GCC 4.6 detects variables that get assined to, but never used later.
Also removes some remmed-out lines that become invalid.
(closes issue ASTERISK-18336)
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>,
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333509
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Jonathan Rose [Fri, 26 Aug 2011 16:38:37 +0000 (16:38 +0000)]
Merged revisions 333410 via svnmerge from
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r333410 | jrose | 2011-08-26 11:28:03 -0500 (Fri, 26 Aug 2011) | 19 lines
Merged revisions 333378 via svnmerge from
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r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines
[patch] Buddies are always auto-registered when processing the roster
Reporter said autoregister flag was ignored for registering 'buddies' which
had a subscription to us. Verified that this was the case and observed how
the patch addressed this and made sure it didn't break anything.
(closes issue ASTERISK-14233)
Reported by: Simon Arlott
Patches:
asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott
Tested by: Jonathan Rose
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Matthew Jordan [Fri, 26 Aug 2011 16:12:13 +0000 (16:12 +0000)]
Merged revisions 333370 via svnmerge from
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r333370 | mjordan | 2011-08-26 10:58:37 -0500 (Fri, 26 Aug 2011) | 26 lines
Merged revisions 333339 via svnmerge from
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r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) | 20 lines
Bug fixes for voicemail user emailsubject / emailbody.
This code change fixes a few issues with the voicemail user override of
emailbody and emailsubject, including escaping the strings, potential memory
leaks, and not overriding the voicemail defaults. Revision 325877 fixed this
for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent
check-in prevented 325877 from being applied to 10. This check-in resolves
both issues, and applies the changes to 1.8, 10, and trunk.
(closes issue ASTERISK-16781)
Reported by: Sebastien Couture
Tested by: mjordan
(closes issue ASTERISK-16795)
Reported by: mdeneen
Tested by: mjordan
Review: https://reviewboard.asterisk.org/r/1374
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Jonathan Rose [Thu, 25 Aug 2011 19:13:23 +0000 (19:13 +0000)]
Merged revisions 333266 via svnmerge from
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r333266 | jrose | 2011-08-25 14:00:05 -0500 (Thu, 25 Aug 2011) | 20 lines
Merged revisions 333265 via svnmerge from
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r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines
Segfault when publishing device states via XMPP and not connected
When using publishing device state with res_jabber, Asterisk will attempt
to send a device state using the unconnected client using iks_send_raw
and crash. This patch checks the validity of the connection before
attempting to send the device state.
(closes issue ASTERISK-18078)
Reported by: Michael L. Young
Patches:
res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young
Tested by: Jonathan Rose
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Jason Parker [Thu, 25 Aug 2011 19:01:39 +0000 (19:01 +0000)]
Merged revisions 333268 via svnmerge from
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r333268 | qwell | 2011-08-25 14:01:18 -0500 (Thu, 25 Aug 2011) | 9 lines
Merged revisions 333267 via svnmerge from
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r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) | 2 lines
Fix for DESTDIR spaces patch.
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Jason Parker [Thu, 25 Aug 2011 15:30:46 +0000 (15:30 +0000)]
Merged revisions 333203 via svnmerge from
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r333203 | qwell | 2011-08-25 10:29:56 -0500 (Thu, 25 Aug 2011) | 15 lines
Merged revisions 333201 via svnmerge from
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r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines
Fix installation into directories containing spaces.
This also vastly simplifies the logic in sounds/Makefile
(Closes issue ASTERISK-18290)
Reported by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/1379/
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Jason Parker [Wed, 24 Aug 2011 19:27:15 +0000 (19:27 +0000)]
Fix typo from r333070
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333159
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Matthew Nicholson [Wed, 24 Aug 2011 16:52:56 +0000 (16:52 +0000)]
Merged revisions 333115 via svnmerge from
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r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug 2011) | 4 lines
Changed the "timeout" option to "gwtimeout".
ASTERISK-18219
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Olle Johansson [Wed, 24 Aug 2011 09:17:48 +0000 (09:17 +0000)]
Formatting changes - Removing some red white space and adding some curly brackets.
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Olle Johansson [Wed, 24 Aug 2011 09:12:23 +0000 (09:12 +0000)]
Add documentation for new manager event in chan_local
AST-17623
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333071
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Olle Johansson [Wed, 24 Aug 2011 09:09:53 +0000 (09:09 +0000)]
Add manager event for local channel semi-bridge
(issue AST-17623)
Review: https://reviewboard.asterisk.org/r/1154
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333070
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Richard Mudgett [Tue, 23 Aug 2011 18:17:52 +0000 (18:17 +0000)]
Merged revisions 333011 via svnmerge from
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r333011 | rmudgett | 2011-08-23 13:15:49 -0500 (Tue, 23 Aug 2011) | 19 lines
Merged revisions 333010 via svnmerge from
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r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) | 12 lines
Memory Leak in app_queue
The patch that was committed in the 1.6.x versions of Asterisk for
ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8
but the other is. queue_leak.patch fixes the portion applicable to 1.8.
(closes issue ASTERISK-18265)
Reported by: Fred Schroeder
Patches:
queue_leak.patch (license #5049) patch uploaded by mmichelson
Tested by: Thomas Arimont
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Richard Mudgett [Mon, 22 Aug 2011 21:25:11 +0000 (21:25 +0000)]
Merged revisions 332940 via svnmerge from
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r332940 | rmudgett | 2011-08-22 16:23:40 -0500 (Mon, 22 Aug 2011) | 14 lines
Merged revisions 332939 via svnmerge from
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r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) | 7 lines
Minor code optimizations.
* Simplify ast_category_browse() logic for easier understanding.
* Remove dead code in ast_variable_delete() and simplify some of its
logic.
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Richard Mudgett [Mon, 22 Aug 2011 20:01:30 +0000 (20:01 +0000)]
Merged revisions 332875,332878 via svnmerge from
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r332875 | rmudgett | 2011-08-22 14:41:03 -0500 (Mon, 22 Aug 2011) | 1 line
Fix merge property.
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r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011) | 25 lines
Merged revisions 332874 via svnmerge from
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r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) | 18 lines
Reference leaks in app_queue.
* Fixed load_realtime_queue() leaking a queue reference when it overwrites
q when processing a realtime queue.
(issue ASTERISK-18265)
* Make join_queue() unreference the queue returned by
load_realtime_queue() when it is done with the pointer. The
load_realtime_queue() returns a reference to the just loaded realtime
queue.
* Fixed queues container reference leak in queues_data_provider_get().
* queue_unref() should not return q that was just unreferenced.
* Made logic in __queues_show() and queues_data_provider_get() when
calling load_realtime_queue() easier to understand.
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Paul Belanger [Mon, 22 Aug 2011 19:56:00 +0000 (19:56 +0000)]
Merged revisions 332877 via svnmerge from
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r332877 | pabelanger | 2011-08-22 15:43:33 -0400 (Mon, 22 Aug 2011) | 13 lines
Merged revisions 332876 via svnmerge from
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r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) | 6 lines
Revert previous commit
It seems google is still making changes to the protocol.
(issue ASTERISK-18301)
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Richard Mudgett [Mon, 22 Aug 2011 19:52:30 +0000 (19:52 +0000)]
Fix merge 10 branch merge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332879
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Matthew Jordan [Mon, 22 Aug 2011 19:19:44 +0000 (19:19 +0000)]
Merged revisions 332817 via svnmerge from
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r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines
Review: https://reviewboard.asterisk.org/r/1364/
This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite.
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Richard Mudgett [Mon, 22 Aug 2011 18:33:27 +0000 (18:33 +0000)]
Merged revisions 332830 via svnmerge from
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r332830 | rmudgett | 2011-08-22 13:32:09 -0500 (Mon, 22 Aug 2011) | 15 lines
Merged revisions 332816 via svnmerge from
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r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines
Memory leaks in realtime_multi_xxx() when database access returns error.
* Fix realtime_multi_pgsql() configuration memory leak when the database
access returns an error.
* Fix realtime_multi_odbc() configuration category use after free when the
database access returns an error.
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Richard Mudgett [Mon, 22 Aug 2011 17:12:16 +0000 (17:12 +0000)]
Merged revisions 332761 via svnmerge from
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r332761 | rmudgett | 2011-08-22 12:05:35 -0500 (Mon, 22 Aug 2011) | 22 lines
Merged revisions 332759 via svnmerge from
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r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) | 15 lines
Memory leak reading realtime database variable list.
Calling ast_load_realtime() can leak the last list node if the read list
only contains empty variable value items.
* Fixed list filter loop in ast_load_realtime() to delete the list node
immediately instead of the next time through the loop. The next time
through the loop may not happen if the node to delete is the last in the
list.
(issue ASTERISK-18277)
(issue ASTERISK-18265)
Patches:
jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett
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Jonathan Rose [Mon, 22 Aug 2011 17:05:14 +0000 (17:05 +0000)]
Add option for logging congested calls as CONGESTION instead of NO_ANSWER in CDR
This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending
with congestion in a way that is unique from other unanswered calls.
(closes issue ASTERISK-14842)
Reported by: Alec Davis
Patches:
cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760
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Matthew Nicholson [Mon, 22 Aug 2011 16:31:59 +0000 (16:31 +0000)]
Merged revisions 332756 via svnmerge from
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r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug 2011) | 4 lines
add a way to disable and/or modify the gateway timeout
ASTERISK-18219
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Paul Belanger [Sun, 21 Aug 2011 14:34:48 +0000 (14:34 +0000)]
Merged revisions 332700 via svnmerge from
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r332700 | pabelanger | 2011-08-21 10:33:23 -0400 (Sun, 21 Aug 2011) | 12 lines
Merged revisions 332699 via svnmerge from
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r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) | 5 lines
Fix outgoing calls in chan_gtalk
(closes issue ASTERISK-18301)
Reported by: az1324
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Kinsey Moore [Fri, 19 Aug 2011 20:00:19 +0000 (20:00 +0000)]
Merged revisions 332654 via svnmerge from
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r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) | 8 lines
Make CONFBRIDGE_INFO behave more nicely
CONFBRIDGE_INFO doesn't behave as well in edge cases as MEETME_INFO. With this
patch, CONFBRIDGE_INFO should behave in a much more reasonable manner when
presented with invalid conferences and keywords.
Review: https://reviewboard.asterisk.org/r/1359/
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Richard Mudgett [Fri, 19 Aug 2011 17:24:56 +0000 (17:24 +0000)]
Fix infinite loop releasing the same memory in ldap_loadentry().
* Fixed memory leak of vars in ldap_loadentry().
* Fixed potential NULL ptr dereference of vars in ldap_loadentry().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332615
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Terry Wilson [Thu, 18 Aug 2011 21:39:04 +0000 (21:39 +0000)]
Merged revisions 332560 via svnmerge from
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r332560 | twilson | 2011-08-18 16:34:04 -0500 (Thu, 18 Aug 2011) | 12 lines
Merged revisions 332559 via svnmerge from
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r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 5 lines
Fix possible error on stringification of IPv4-mapped addrs
The FreeBSD netsock2 test has been failing for a while. We were
pasing sa->len to getnameinfo instead of sa_tmp->len.
ASTERISK-18289
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Kinsey Moore [Thu, 18 Aug 2011 19:30:04 +0000 (19:30 +0000)]
Merged revisions 332504 via svnmerge from
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r332504 | kmoore | 2011-08-18 14:29:15 -0500 (Thu, 18 Aug 2011) | 15 lines
Merged revisions 332503 via svnmerge from
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r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines
CRC4 in "dahdi show status" gives wrong impression to T1 users
Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in
more situations without confusing users, especially since T1 lines use CRC6
instead of CRC4.
(closes issue AST-471)
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Tilghman Lesher [Thu, 18 Aug 2011 14:49:28 +0000 (14:49 +0000)]
Merged revisions 332447 via svnmerge from
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r332447 | tilghman | 2011-08-18 09:48:40 -0500 (Thu, 18 Aug 2011) | 9 lines
Merged revisions 332446 via svnmerge from
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r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 Aug 2011) | 2 lines
Move BETTER_BACKTRACES out of development mode, as it's useful when DEBUG_THREADS is enabled.
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Tilghman Lesher [Wed, 17 Aug 2011 19:30:50 +0000 (19:30 +0000)]
Merged revisions 332369 via svnmerge from
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r332369 | tilghman | 2011-08-17 14:24:59 -0500 (Wed, 17 Aug 2011) | 17 lines
Merged revisions 332355 via svnmerge from
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r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 10 lines
Re-add support for spaces in pathnames, including now spaces in DESTDIR.
This was initially added to 1.8 prior to release, primarily to support the
standard paths on Mac OS X, but was partially reverted recently in Subversion,
due to the lack of support for spaces in DESTDIR. This commit restores support
for the standard paths on Mac OS X, and also includes support for spaces in
DESTDIR.
(closes issue ASTERISK-18290)
Reported by: pabelanger
Review: https://reviewboard.asterisk.org/r/1326/
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Terry Wilson [Wed, 17 Aug 2011 18:31:39 +0000 (18:31 +0000)]
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r332321 | twilson | 2011-08-17 13:09:49 -0500 (Wed, 17 Aug 2011) | 17 lines
Merged revisions 332320 via svnmerge from
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r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) | 10 lines
Don't read from a disarmed or invalid timerfd
Numerous isues have been reported for deadlocks that are caused by
a blocking read in res_timing_timerfd on a file descriptor that will
never be written to. This patch adds some checks to make sure that
the timerfd is both valid and armed before calling read().
Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486,
AST-495, AST-507 and possibly others.
Review: https://reviewboard.asterisk.org/r/1361/
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Richard Mudgett [Wed, 17 Aug 2011 16:18:27 +0000 (16:18 +0000)]
Merged revisions 332265 via svnmerge from
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r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines
Merged revisions 332264 via svnmerge from
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r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines
Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
France Telecom brings layer 2 and layer 1 down on BRI lines when the line
is idle. When layer 1 goes down Asterisk cannot make outgoing calls and
the HA8 and HB8 cards also get IRQ misses.
The inability to make outgoing calls is because the line is in red alarm
and Asterisk will not make calls over a line it considers unavailable.
The IRQ misses for the HA8 and HB8 card are because the hardware is
switching clock sources from the line which just brought layer 1 down to
internal timing.
There is a DAHDI option for the B410P card to not tell Asterisk that layer
1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp
teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards:
"modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear
up the IRQ misses when the telco brings layer 1 down.
* Add layer 2 persistence option to customize the layer 2 behavior on BRI
PTMP lines. The new option has three settings: 1) Use libpri default
layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer
brings it down. 3) Leave layer 2 down when the peer brings it down.
Layer 2 will be brought up as needed for outgoing calls.
JIRA AST-598
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Paul Belanger [Tue, 16 Aug 2011 20:15:23 +0000 (20:15 +0000)]
Merged revisions 332177 via svnmerge from
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r332177 | pabelanger | 2011-08-16 16:11:49 -0400 (Tue, 16 Aug 2011) | 11 lines
Merged revisions 332176 via svnmerge from
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r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug 2011) | 4 lines
Flag test modules as 'core'
Review: https://reviewboard.asterisk.org/r/1369/
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Jonathan Rose [Tue, 16 Aug 2011 17:53:23 +0000 (17:53 +0000)]
Merged revisions 332119 via svnmerge from
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r332119 | jrose | 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) | 23 lines
Merged revisions 332118 via svnmerge from
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r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines
ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on whichever mailbox
triggered the mwi event. Now all of them get counted regardless. Also fixes a bug
involving parsing of the mailbox option in sip.conf so that trailing and leading
spaces before/after commas are trimmed.
(closes issue ASTERISK-18067)
Reported by: aragon
(closes issue ASTERISK-15479)
Reported by: Ben Winslow
Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow
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Richard Mudgett [Tue, 16 Aug 2011 17:23:08 +0000 (17:23 +0000)]
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r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines
Merged revisions 332100 via svnmerge from
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r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines
Fix multiple parking issues.
JIRA ASTERISK-17183
Multi-parkinglot directs calls to wrong parkinglot.
JIRA ASTERISK-17870
Cannot retrieve parked calls.
JIRA ASTERISK-17430
ParkedCall() with no extension should pickup first available call and does not.
JIRA AST-576
Issues with parking lots
* Removed searching for parking lots by extension. Parking lots can only
be found by the parking lot name since parking lot access extensions and
spaces are not guaranteed to be unique.
* Added parking_lot_name option to the Park and ParkedCall applications.
Updated documentation for Park and ParkedCall applications.
* Add parkext_exclusive configuration option to make parking entry
extensions specify which parking lot they access.
(closes issue ASTERISK-17183)
Reported by: David Cabrejos
Tested by: rmudgett, David Cabrejos
(closes issue ASTERISK-17870)
Reported by: Remi Quezada
(closes issue ASTERISK-17430)
Reported by: Philippe Lindheimer
JIRA ASTERISK-17452
Parking_offset not used
JIRA AST-624
'next' setting for findslot does nothing
* Reimplemented since findslot feature option broken by -r114655.
(closes issue ASTERISK-17452)
Reported by: David Woolley
Tested by: rmudgett
JIRA ASTERISK-15792
Dialplan continues execution after transfer to park.
This happens for DTMF attended transfer, DTMF blind transfer, and DTMF
one-touch-parking if the party initiating these features also initiated
the call.
* Fixed the return code from the affected builtin features when parking a
call.
(closes issue ASTERISK-15792)
Reported by: Mat Murdock
Tested by: rmudgett, twilson
JIRA AST-607
The courtesytone is not playing to the expected call when picking up a
parked call.
This is mostly a documentation problem. However, the option is not reset
to the default when features.conf is reloaded.
* Updated features.conf.sample documentation for courtesytone and
parkedplay options.
* Reset the parkedplay option to default when features.conf is reloaded.
JIRA AST-615
AMI Park action followed by features reload results in orphaned channels
in parking lot.
* Reloading features.conf will not touch parking lots that have calls
still parked in them. Reload again at a later time.
Misc additional fixes:
* Added unit test for parking lot dialplan usage checking.
* Made update connected line when a parked call is retrieved from a
parking lot.
* Made retrieved parked call stop ringing or MOH depending upon how the
call was waiting in the parking lot.
* Made CLI "features show" indicate if the parking lot is enabled for use.
* Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to
specify the parking lot access extension.
* Made AMI ParkedCalls action ParkedCall events have a Parkinglot header.
* Made AMI ParkedCalls action ParkedCallsComplete event have a Total
header.
* Fixed potential deadlock from AMI Park action holding channel locks
while calling masq_park_call().
* Fixed several places where ast_strdupa() were used inside of loops.
(Mostly fixed by refactoring the loop body into its own function.)
* Fixed copy_parkinglot() copying too much from the source parking lot.
Extracted the parking lot configuration settings into struct
parkinglot_cfg.
* Refactored courtesytone playing code to put the channel not playing the
tone in autoservice.
* Fix when pbx-parkingfailed is played that the other channel is put in
autoservice if it exists.
* Fixed parkinglot reference leak in parked_call_exec() error paths.
* Fixed parkinglot_unref() use of parkinglot after it was unreffed.
* Made destroy the struct ast_parkinglot parkings lock when done.
* Refactored the features.conf parking lot configuration code to eliminate
redundancy.
* Fixed feature reload to better protect parking lots.
* Fixed parking lot container reference leak in handle_parkedcalls().
* Fixed the total count in handle_parkedcalls().
Review: https://reviewboard.asterisk.org/r/1358/
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Matthew Nicholson [Tue, 16 Aug 2011 15:21:18 +0000 (15:21 +0000)]
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r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) | 2 lines
fix a code comment
AST-580
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Matthew Nicholson [Tue, 16 Aug 2011 15:17:56 +0000 (15:17 +0000)]
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r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) | 2 lines
Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES
AST-580
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Matthew Nicholson [Tue, 16 Aug 2011 15:10:18 +0000 (15:10 +0000)]
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r332027 | mnicholson | 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) | 9 lines
Merged revisions 332026 via svnmerge from
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r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines
use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option
AST-580
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Olle Johansson [Tue, 16 Aug 2011 14:47:38 +0000 (14:47 +0000)]
Formatting changes while working with DTMF...
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Matthew Nicholson [Tue, 16 Aug 2011 14:41:23 +0000 (14:41 +0000)]
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r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from
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r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
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Richard Mudgett [Mon, 15 Aug 2011 17:36:47 +0000 (17:36 +0000)]
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r331956 | rmudgett | 2011-08-15 12:35:03 -0500 (Mon, 15 Aug 2011) | 20 lines
Merged revisions 331955 via svnmerge from
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r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines
Fix some minor chan_dahdi config load issues.
* Address chan_dahdi.conf dahdichan option todo item about needing line
number.
* Make ignore_failed_channels option also apply to dahdichan option.
* Don't attempt to create a default pseudo channel if the chan_dahdi.conf
channel/channels option is not allowed.
* Add a similar check for dahdichan in normal chan_dahdi.conf sections as
is done in users.conf.
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Paul Belanger [Mon, 15 Aug 2011 15:24:55 +0000 (15:24 +0000)]
Merged revisions 331894 via svnmerge from
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r331894 | pabelanger | 2011-08-15 11:22:45 -0400 (Mon, 15 Aug 2011) | 12 lines
Merged revisions 331886 via svnmerge from
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r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug 2011) | 5 lines
Fix noisy message when briding channels
(closes issue ASTERISK-18270)
Reported by: Federico Alves
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David Vossel [Mon, 15 Aug 2011 15:15:43 +0000 (15:15 +0000)]
Merged revisions 331868 via svnmerge from
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r331868 | dvossel | 2011-08-15 10:14:13 -0500 (Mon, 15 Aug 2011) | 12 lines
Merged revisions 331867 via svnmerge from
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r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) | 6 lines
Fixes locking inversion issues present in the handling of the sip REFER method.
(closes issue ASTERISK-18082)
Reported by: James Van Vleet
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Olle Johansson [Mon, 15 Aug 2011 13:27:06 +0000 (13:27 +0000)]
Formatting guideline fixes
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Matthew Nicholson [Fri, 12 Aug 2011 19:06:10 +0000 (19:06 +0000)]
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r331775 | mnicholson | 2011-08-12 14:03:31 -0500 (Fri, 12 Aug 2011) | 17 lines
Merged revisions 331774 via svnmerge from
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r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug 2011) | 11 lines
Unlock the channel before calling update_queue.
Holding the channel lock when calling update_queue which attempts to lock the
queue lock can cause a deadlock. This deadlock involves the following chain:
1. hold chan lock -> wait queue lock
2. hold queue lock -> wait agent list lock
3. hold agent list lock -> wait chan list lock
4. hold chan list lock -> wait chan lock
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Richard Mudgett [Fri, 12 Aug 2011 19:01:04 +0000 (19:01 +0000)]
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r331772 | rmudgett | 2011-08-12 13:59:45 -0500 (Fri, 12 Aug 2011) | 15 lines
Merged revisions 331771 via svnmerge from
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r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines
Suppress warning message when using DAHDITransfer or DAHDIHangup.
* The fake event should only be processed by the channel that currently
owns the private and not the associated call waiting or 3-way channel.
JIRA AST-620
JIRA SWP-3616
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Jonathan Rose [Fri, 12 Aug 2011 18:03:29 +0000 (18:03 +0000)]
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r331644 | jrose | 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) | 9 lines
Merged revisions 331635 via svnmerge from
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r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line
Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme
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Richard Mudgett [Fri, 12 Aug 2011 17:56:46 +0000 (17:56 +0000)]
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r331715 | rmudgett | 2011-08-12 12:54:47 -0500 (Fri, 12 Aug 2011) | 29 lines
Merged revisions 331714 via svnmerge from
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r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines
AMI actions DAHDIHangup and DAHDITransfer have no effect.
The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI
channel. These two AMI actions are highly specialized to analog channels
and appear to make the channel behave like a jack port for headsets.
* Made the faked DAHDI event get processed before a normal media stream
read in dahdi_read() instead of trying to trigger an exception read by
setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long
ago that changed how AST_FLAG_EXCEPTION is processed in the core.
Unfortunately, the faked DAHDI events no longer worked when that happened.
* Updated the DAHDI AMI action documentation for the following actions:
DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff,
DAHDIShowChannels, and DAHDIRestart.
* Made use sscanf() instead of atoi() for better error checking of the
DAHDIChannel header string.
JIRA AST-620
JIRA SWP-3616
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Terry Wilson [Fri, 12 Aug 2011 16:32:24 +0000 (16:32 +0000)]
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r331659 | twilson | 2011-08-12 11:31:21 -0500 (Fri, 12 Aug 2011) | 11 lines
Merged revisions 331658 via svnmerge from
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r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011) | 4 lines
Fix netsock2 multiple zero-expansion test
Remove erroneous single bracket.
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Kinsey Moore [Fri, 12 Aug 2011 16:22:45 +0000 (16:22 +0000)]
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r331654 | kmoore | 2011-08-12 11:21:37 -0500 (Fri, 12 Aug 2011) | 19 lines
Merged revisions 331649 via svnmerge from
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r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) | 12 lines
Logger does not warn of failure to open logging channels
Currently, logger only prints an error message to stderr when it fails to open
a logger channel where many users will not see it because the logger lock is
held. The alternative provided by this patch is to log the error to all
attached consoles in the hopes that it will be easier to see. Additionally,
this patch prevents the failed logger channel from being added to the list
where it would silently fail on each call to the Asterisk logger.
(closes issue ASTERISK-16231)
Review: https://reviewboard.asterisk.org/r/1338
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Jason Parker [Thu, 11 Aug 2011 21:55:48 +0000 (21:55 +0000)]
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r331579 | qwell | 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) | 13 lines
Merged revisions 331578 via svnmerge from
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r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines
Use proper values for 64-bit option flags.
Also, reusing bits es no bueno, so change the value of a duplicate.
(issue ASTERISK-18239)
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Richard Mudgett [Thu, 11 Aug 2011 21:44:08 +0000 (21:44 +0000)]
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r331576 | rmudgett | 2011-08-11 16:42:21 -0500 (Thu, 11 Aug 2011) | 16 lines
Merged revisions 331575 via svnmerge from
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r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines
Segfault in shell_helper in func_shell.c.
The return value of popen() was not checked for failure to open.
(closes issue ASTERISK-18109)
JIRA SWP-3633
Reported by: Michael Myles
Tested by: rmudgett
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Kinsey Moore [Wed, 10 Aug 2011 22:24:38 +0000 (22:24 +0000)]
Merged revisions 331518 via svnmerge from
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r331518 | kmoore | 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) | 17 lines
Merged revisions 331517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines
SIP Notify via AMI or CLI leaks SIP PVTs
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
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Richard Mudgett [Wed, 10 Aug 2011 20:51:07 +0000 (20:51 +0000)]
Merged revisions 331462 via svnmerge from
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r331462 | rmudgett | 2011-08-10 15:41:35 -0500 (Wed, 10 Aug 2011) | 37 lines
Merged revisions 331461 via svnmerge from
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r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) | 30 lines
Output of queue log not started until logger reloaded.
ASTERISK-15863 caused a regression with queue logging. The output of the
queue log is not started until the logger configuration is reloaded.
* Queue log initialization is completely delayed until the first message
is posted to the queue log system. Including the initial opening of the
queue log file.
* Fixed rotate_file() ROTATE strategy to give the file just rotated out to
the configured exec function after rotate. Just like the other strategies.
* Fixed logger reload to always post the queue reload entry instead of
just if there is a queue log file.
* Refactored some code to eliminate some redundancy and to reduce stack
utilization.
(closes issue ASTERISK-17036)
JIRA SWP-2952
Reported by: Juan Carlos Valero
Patches:
jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
(closes issue ASTERISK-18208)
Reported by: Christian Pinedo
Review: https://reviewboard.asterisk.org/r/1333/
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Richard Mudgett [Wed, 10 Aug 2011 19:08:22 +0000 (19:08 +0000)]
Merged revisions 331420 via svnmerge from
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r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011) | 2 lines
Make sure feature_request_and_dial() initializes outstate if passed in.
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Richard Mudgett [Wed, 10 Aug 2011 18:27:16 +0000 (18:27 +0000)]
Merged revisions 331418 via svnmerge from
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r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) | 6 lines
Revert -r318141. It was a band-aid that only partially fixed parking.
A better fix is on reviewboard review 1358.
(issue ASTERISK-17374)
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Jonathan Rose [Wed, 10 Aug 2011 15:45:57 +0000 (15:45 +0000)]
SIP display-name needed to be empty for Avaya IP500
In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present. Instead, it will send no display name in that case.
(closes issue ASTERISK-16198)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/1341/
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Kinsey Moore [Wed, 10 Aug 2011 13:49:31 +0000 (13:49 +0000)]
Merged revisions 331316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r331316 | kmoore | 2011-08-10 08:48:41 -0500 (Wed, 10 Aug 2011) | 15 lines
Merged revisions 331315 via svnmerge from
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r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | 8 lines
AMI action ModuleReload returns Error if Module: missing or empty
An empty string was not being checked for properly causing identification of
the module to be reloaded to fail and return an Error with message
"No such module."
(closes issue AST-616)
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Richard Mudgett [Tue, 9 Aug 2011 23:17:13 +0000 (23:17 +0000)]
Merged revisions 331265 via svnmerge from
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r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines
Merged revisions 331248 via svnmerge from
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r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines
Misc minor items found in code.
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
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Alexandr Anikin [Tue, 9 Aug 2011 17:12:27 +0000 (17:12 +0000)]
Merged revisions 331147,331200 via svnmerge from
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r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, 09 Aug 2011) | 11 lines
Merged revisions 331146 via svnmerge from
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r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines
move ast_cond_signal for admitted call after all data filled/freed
clear all log channels by pointed number not only first
free allocated callToken in ooh323_answer
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r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, 09 Aug 2011) | 9 lines
Setup IP proto version for call in GK mode
Added additional check for IP semantics before parse destination
by ast_parse_args due to it can parse numeric as IP.
(closes issue ASTERISK-18218)
Reported by: slesru
Patch: ASTERISK-18218.patch
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Kinsey Moore [Tue, 9 Aug 2011 17:08:33 +0000 (17:08 +0000)]
Allow ENUM query functions to report lookup errors
The ENUM dialplan functions do not report DNS query errors properly. It is
useful to differentiate between failed query (e.g. non-existent domain) vs. no
data records of the appropriate type. This is required to make overlapped
dialing work.
(closes issue ASTERISK-13769)
Review: https://reviewboard.asterisk.org/r/1355/
Patch-by: Timo Teras
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Jason Parker [Tue, 9 Aug 2011 16:02:45 +0000 (16:02 +0000)]
Merged revisions 331143 via svnmerge from
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r331143 | qwell | 2011-08-09 10:59:54 -0500 (Tue, 09 Aug 2011) | 9 lines
Merged revisions 331142 via svnmerge from
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r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line
Regenerate asterisk man page from sgml.
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Jason Parker [Tue, 9 Aug 2011 15:53:26 +0000 (15:53 +0000)]
Merged revisions 331139 via svnmerge from
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r331139 | qwell | 2011-08-09 10:50:07 -0500 (Tue, 09 Aug 2011) | 19 lines
Merged revisions 306999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines
Documentation Updates
Note default polling setting in voicemail.conf
Add missing config to asterisk.conf
Update manpage
(issue #16505)
Reported by: tzafrir
Patches:
asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir
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Jason Parker [Tue, 9 Aug 2011 15:51:38 +0000 (15:51 +0000)]
Merged revisions 331138 via svnmerge from
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r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line
Revert merge of r306999, due to merge conflict.
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Terry Wilson [Mon, 8 Aug 2011 22:59:45 +0000 (22:59 +0000)]
Merged revisions 331097 via svnmerge from
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r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines
Bump the AMI protocol version to 1.2
As a result of converting Unlink events that were missed in the AMI
1.1 update to Bridge events, the AMI protocol version is being incremented.
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Terry Wilson [Mon, 8 Aug 2011 21:16:25 +0000 (21:16 +0000)]
Merged revisions 331041 via svnmerge from
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r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines
Replace AMI Unlink events with Bridge events
A previous update converted some of the Link and Unlink events to
Bridge events, but a couple of Unlink events were missed. This patch
rectifies the situation.
(closes issues ASTERISK-17455)
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Kinsey Moore [Mon, 8 Aug 2011 20:54:14 +0000 (20:54 +0000)]
Merged revisions 331039 via svnmerge from
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r331039 | kmoore | 2011-08-08 15:53:30 -0500 (Mon, 08 Aug 2011) | 18 lines
Merged revisions 331038 via svnmerge from
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r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines
In-queue MOH stops after a periodic announcement
If the seek value is past the end of file when resuming G.722 MOH, MOH will
cease to function for the duration of the MOH session through all starts and
stops until saved state is cleared. Adjusting the code to guarantee a single
valid read (which is already assumed) fixes the bug.
(closes issue ASTERISK-18077)
Review: https://reviewboard.asterisk.org/r/1328/
Tested-by: Jonathan Rose <jrose@digium.com>
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Kinsey Moore [Mon, 8 Aug 2011 20:28:20 +0000 (20:28 +0000)]
Log queue member name when state_interface is set for ADDMEMBER and REMOVEMEMBER events
app_queue logs the events ADDMEMBER and REMOVEMEMBER with the agent field set
to the interface value rather than the membername value when a member is added
with a state_interface value set. However all other member related queue
events are logged with the membername when a state_interface is set. This
patch makes these fields optionally more consistent and correct.
(closes issue ASTERISK-14769)
Review: https://reviewboard.asterisk.org/r/1286
Patch-by: Jamuel Starkey
Tested-by: Kinsey Moore <kmoore@digium.com>
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Kinsey Moore [Mon, 8 Aug 2011 15:00:26 +0000 (15:00 +0000)]
app_queue: Add StateInterface to output of "queue show" and "QueueStatus"
This patch adds the state_interface of the queue member struct to the output
of "queue show" (CLI command) and "QueueStatus" (AMI action) when displaying
relevant queue member information. For the AMI event message the variable
StateInterface has been added.
(closes issue ASTERISK-18071)
Review: https://reviewboard.asterisk.org/r/1300/
Patch-by: Jamuel Starkey
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David Vossel [Fri, 5 Aug 2011 15:57:06 +0000 (15:57 +0000)]
Merged revisions 330940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011) | 2 lines
The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly.
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