Mark Michelson [Wed, 17 Dec 2008 17:53:37 +0000 (17:53 +0000)]
And actually assign the function to a pointer...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165143
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Mark Michelson [Wed, 17 Dec 2008 17:52:50 +0000 (17:52 +0000)]
Use the create_vm_state_from_user function in a place where
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.
This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165142
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Kevin P. Fleming [Wed, 17 Dec 2008 16:41:18 +0000 (16:41 +0000)]
ignore this copied file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165108
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Steve Murphy [Wed, 17 Dec 2008 05:04:56 +0000 (05:04 +0000)]
A possibly "horrible fix" for a "horribly broken"
situation.
As stuff shifts around in the asterisk code, the
miscellaneous inclusions from the standalone stuff
gets broken. There's no easy fix for this situation.
I made sure that everything in utils builds without
problem ***AND*** that aelparse runs the regressions
correctly with the following make menuselect options
both on and off:
DONT_OPTIMIZE
DEBUG_THREADS
DEBUG_CHANNEL_LOCKS
MALLOC_DEBUG
MTX_PROFILE
DEBUG_SCHEDULER
DEBUG_THREADLOCALS
DETECT_DEADLOCKS
CHANNEL_TRACE
I think from now on, I'm going to #undef
all these features in the various utils native
files; I guess I could do the same for the
copied-in files, surrounded by STANDALONE ifdef.
A standalone isn't going to care about threads,
mutexes, etc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165071
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Steve Murphy [Wed, 17 Dec 2008 02:32:58 +0000 (02:32 +0000)]
fixed the regressions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165039
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Mark Michelson [Tue, 16 Dec 2008 23:06:04 +0000 (23:06 +0000)]
Merged revisions 164977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines
After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164978
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Tilghman Lesher [Tue, 16 Dec 2008 22:57:17 +0000 (22:57 +0000)]
Add timezone to the possible fields in a timespec.
(closes issue #14028)
Reported by: mostyn
Patches:
timezone-v2.patch uploaded by mostyn (license 398)
(with additional code guideline fixes and a memory leak fix by me - license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164976
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Jeff Peeler [Tue, 16 Dec 2008 22:45:39 +0000 (22:45 +0000)]
(closes issue #13669)
Reported by: pj
Delete file recording if recording terminated from a hangup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164942
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Terry Wilson [Tue, 16 Dec 2008 22:31:47 +0000 (22:31 +0000)]
Make a note of the feature request in bug #11157 as per the reporter and oej, and suspend the bug since no one seems to be keen on implementing it any time soon.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164941
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Russell Bryant [Tue, 16 Dec 2008 21:39:15 +0000 (21:39 +0000)]
Merged revisions 164881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) | 9 lines
Fix an issue where DEBUG_THREADS may erroneously report that a thread
is exiting while holding a lock.
If the last lock attempt was a trylock, and it failed, it will still be in the
list of locks so that it can be reported.
(closes issue #13219)
Reported by: pj
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Russell Bryant [Tue, 16 Dec 2008 21:12:49 +0000 (21:12 +0000)]
Merged revisions 164876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines
Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.
This is a bug I noticed while looking at the code for app_macro. This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched. (I hate this return code with a passion, by the way.)
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Russell Bryant [Tue, 16 Dec 2008 20:49:25 +0000 (20:49 +0000)]
Fix build issues on Linux after sysinfo related changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164821
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Joshua Colp [Tue, 16 Dec 2008 20:47:31 +0000 (20:47 +0000)]
Qualify trumps poke per lmadsen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814
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Joshua Colp [Tue, 16 Dec 2008 20:42:33 +0000 (20:42 +0000)]
Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time.
(closes issue #13217)
Reported by: cervajs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809
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Russell Bryant [Tue, 16 Dec 2008 20:41:51 +0000 (20:41 +0000)]
Merged revisions 164806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines
Add "restart gracefully" to the AMI blacklist of CLI commands.
"module unload" was already identified as a command that can not be used
from the AMI. "restart gracefully" effectively unloads all modules, and will
run in to the same problems.
(closes issue #13894)
Reported by: kernelsensei
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164807
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Michiel van Baak [Tue, 16 Dec 2008 20:08:34 +0000 (20:08 +0000)]
introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl.
(closes issue #13433)
Reported by: mvanbaak
Patches:
2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7)
with two free calls replaced with ast_free based on feedback on reviewboard
Review:
http://reviewboard.digium.com/r/91/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164802
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Steve Murphy [Tue, 16 Dec 2008 20:04:46 +0000 (20:04 +0000)]
(closes issue #14076)
Reported by: toc
Tested by: murf
OK, Well this issue has had its share of flip-flopping.
I found the following:
1. the code in question, in ext_cmp1 in pbx.c, would not
allow two extensions that vary only by any dashes contained
within them, to be defined in the same context.
2. for input dialstrings, dashes are NOT ignored.
So, skipping them when sorting patterns seemed a bit silly.
Thus, you might declare ext 891 in a context, but
if you try dialing 8-9-1, it will NOT match 891.
So, I proposed to remove the code from ext_cmp1 to
skip the spaces and dashes. Just kept us from
declaring 891 and 8-9-1 in the same context,
forcing users to generate otherwise uselessly
obfuscated dialplan code to get the same effect.
Then, I tried out 1.4, and found that:
1. you can declare 891 and 8-9-1 in the
same context!
2. You can't define 891, and have 8-9-1 match
it! Nor can you define 8-9-1, and have 891
match it!
So, it appears that my proposal simply restores
the pbx to behaving as it did in 1.4.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164801
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Tilghman Lesher [Tue, 16 Dec 2008 19:54:11 +0000 (19:54 +0000)]
Set up umask as a possible configuration option.
(closes issue #13753)
Reported by: irroot
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164798
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Russell Bryant [Tue, 16 Dec 2008 17:14:01 +0000 (17:14 +0000)]
Merged revisions 164736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines
Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all. This led to a memory
leak.
Another issue was an invalid argument being provided to the the object_add
API call.
(closes issue #13678)
Reported by: ys
Tested by: Russell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164737
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Joshua Colp [Tue, 16 Dec 2008 16:50:11 +0000 (16:50 +0000)]
Be more detailed about why the include did not get included.
(closes issue #14071)
Reported by: kshumard
Patches:
pbx_config.patch.improvederroroutput.txt uploaded by kshumard (license 92)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164733
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Russell Bryant [Tue, 16 Dec 2008 16:00:29 +0000 (16:00 +0000)]
Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines
Fix a memory leak related to the use of the "setvar" configuration option.
The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.
(closes issue #14037)
Reported by: marvinek
Tested by: russell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164675
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Joshua Colp [Tue, 16 Dec 2008 15:44:28 +0000 (15:44 +0000)]
When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself.
(closes issue #13634)
Reported by: performer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164659
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Steve Murphy [Tue, 16 Dec 2008 15:31:54 +0000 (15:31 +0000)]
Merged revisions 164634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 lines
I added a sentence to clarify why - and ' ' are ignored in patterns
as per bug 14076. Leif says he'll put some stuff about it in the
extensions.conf sample, etc.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164648
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Russell Bryant [Tue, 16 Dec 2008 15:00:27 +0000 (15:00 +0000)]
Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was not needed.
(closes issue #14081)
Reported by: pkempgen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164623
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Russell Bryant [Tue, 16 Dec 2008 14:31:02 +0000 (14:31 +0000)]
Merged revisions 164605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008) | 5 lines
Don't try to change working directory if a directory was not configured.
(closes issue #14089)
Reported by: caspy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164606
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Russell Bryant [Tue, 16 Dec 2008 14:17:45 +0000 (14:17 +0000)]
Fix usage of the DAHDI_VMWI ioctl.
(closes issue #14090)
Reported by: alecdavis
Patches:
chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164602
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Sean Bright [Tue, 16 Dec 2008 01:52:32 +0000 (01:52 +0000)]
Use tables instead of ASCII art. Also change a bit of minor formatting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164565
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Russell Bryant [Mon, 15 Dec 2008 22:25:46 +0000 (22:25 +0000)]
Open a timer before loading configuration so that the trunking configuration option
will take effect.
(closes issue #14082)
Reported by: seandarcy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164525
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Russell Bryant [Mon, 15 Dec 2008 22:22:43 +0000 (22:22 +0000)]
Fix log message to refer to the generic timing interface, not DAHDI specifically
(inspired by issue #14082)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164522
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Russell Bryant [Mon, 15 Dec 2008 21:53:30 +0000 (21:53 +0000)]
Make sure we handle a uint32_t payload in ast_frdup()
(closes issue #14080)
Reported by: fnordian
Patches:
frame.patch uploaded by fnordian (license 110)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164519
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Tilghman Lesher [Mon, 15 Dec 2008 21:17:07 +0000 (21:17 +0000)]
Allow disabling pattern match searches within the Realtime dialplan switch.
(closes issue #13698)
Reported by: fhackenberger
Patches:
20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
Tested by: fhackenberger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164485
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Mark Michelson [Mon, 15 Dec 2008 20:07:03 +0000 (20:07 +0000)]
Add an 'i' option to app_page. This option works the same as
the 'i' options for app_dial and app_queue, in that they will ignore
any attempts by phones to forward the call.
(closes issue #13977)
Reported by: putnopvut
Patches:
page_ignore_forwards.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, acunningham
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164428
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Mark Michelson [Mon, 15 Dec 2008 19:53:29 +0000 (19:53 +0000)]
Merged revisions 164422 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines
Add the deadlock note to ast_spawn_extension as well
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Mark Michelson [Mon, 15 Dec 2008 19:51:24 +0000 (19:51 +0000)]
Merged revisions 164416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines
Add notes to autoservice and pbx doxygen regarding a potential
deadlock scenario so that it is avoided in the future
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164419
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Tilghman Lesher [Mon, 15 Dec 2008 19:48:02 +0000 (19:48 +0000)]
Revert ast_str opacity in chan_sip for now, since something wasn't quite right
in the merge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164417
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Steve Murphy [Mon, 15 Dec 2008 19:42:05 +0000 (19:42 +0000)]
I was getting this warning during a compile
on a 64-bit machine running ubuntu server 8.10,
and gcc-4.3.2:
[CXXi] chan_vpb.ii -> chan_vpb.oo
cc1plus: warnings being treated as errors
In file included from /home/murf/asterisk/trunk/include/asterisk/utils.h:671,
from chan_vpb.cc:46:
/home/murf/asterisk/trunk/include/asterisk/strings.h: In function ‘char* ast_str_truncate(ast_str*, ssize_t)’:
/home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: comparison between signed and unsigned integer expressions
make[1]: *** [chan_vpb.oo] Error 1
make: *** [channels] Error 2
which this fix silences
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164415
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Joshua Colp [Mon, 15 Dec 2008 18:12:24 +0000 (18:12 +0000)]
Merged revisions 164350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines
Do not try to unlock a non-existant channel if the transfer fails.
(closes issue #13800)
Reported by: dwagner
Patches:
asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164351
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Tilghman Lesher [Mon, 15 Dec 2008 18:09:58 +0000 (18:09 +0000)]
When querying for the structure of the CDR table, remove the schema, if it
exists.
(Closes issue #14058)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164349
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Joshua Colp [Mon, 15 Dec 2008 17:46:32 +0000 (17:46 +0000)]
Blocked revisions 164343 via svnmerge
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r164343 | file | 2008-12-15 13:43:59 -0400 (Mon, 15 Dec 2008) | 4 lines
Use autoconf logic to determine whether the system has timersub or not. Do not blindly assume Solaris does not.
(closes issue #13838)
Reported by: ano
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164344
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Joshua Colp [Mon, 15 Dec 2008 17:24:28 +0000 (17:24 +0000)]
Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module.
(closes issue #14079)
Reported by: elguero
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164312
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Russell Bryant [Mon, 15 Dec 2008 17:21:38 +0000 (17:21 +0000)]
Fix a couple more build issues related to ast_str_opaque
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164309
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Russell Bryant [Mon, 15 Dec 2008 16:17:55 +0000 (16:17 +0000)]
When a reload is issued, always process the configuration for dundi.conf.
The reason is that a reload can be used to refresh DNS lookups for defined peers.
Even if the config file hasn't changed, we want to process it for that purpose.
(closes issue #13776)
Reported by: kombjuder
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164272
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Mark Michelson [Mon, 15 Dec 2008 16:16:47 +0000 (16:16 +0000)]
Fix a compile warning and a logic error that could have been bad
for non-realtime queues
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164270
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Mark Michelson [Mon, 15 Dec 2008 16:10:43 +0000 (16:10 +0000)]
Fix up a few issues with regards to queues
* Fix reference counting used in the __queues_show function
* Add code to be sure that the "queue show" command does not
print information for a realtime queue which has been deleted
from the backend
* Add a missing unref to the realtime queue loading function for
the case where a queue is in the module's container but has been
deleted from the realtime backend
(closes issue #14033)
Reported by: cristiandimache
Patches:
14033.patch uploaded by putnopvut (license 60)
Tested by: cristiandimache
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164268
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Joshua Colp [Mon, 15 Dec 2008 15:41:22 +0000 (15:41 +0000)]
Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret.
(closes issue #14073)
Reported by: seandarcy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164257
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Joshua Colp [Mon, 15 Dec 2008 15:21:23 +0000 (15:21 +0000)]
Update to work with new ast_str changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164208
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Joshua Colp [Mon, 15 Dec 2008 15:06:38 +0000 (15:06 +0000)]
Blocked revisions 164204 via svnmerge
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r164204 | file | 2008-12-15 11:05:08 -0400 (Mon, 15 Dec 2008) | 4 lines
Can we try not to assign an unsigned int to -1?
(closes issue #14074)
Reported by: wetwired
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164205
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Russell Bryant [Mon, 15 Dec 2008 14:40:24 +0000 (14:40 +0000)]
Merged revisions 164201 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines
Handle a case where a call can be bridged to a channel that is still ringing.
The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
(closes issue #13747)
Reported by: davidw
Tested by: russell
Review: http://reviewboard.digium.com/r/90/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164203
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Russell Bryant [Mon, 15 Dec 2008 14:40:08 +0000 (14:40 +0000)]
Fix build WRT ast_str_opaque
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164202
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Tilghman Lesher [Sun, 14 Dec 2008 18:16:28 +0000 (18:16 +0000)]
Don't pass a negative to an unsigned type and expect things to work correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164168
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Sean Bright [Sun, 14 Dec 2008 15:26:37 +0000 (15:26 +0000)]
Use a \picture instead of ASCII art.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164137
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Tilghman Lesher [Sat, 13 Dec 2008 23:24:09 +0000 (23:24 +0000)]
Blocked revisions 164082 via svnmerge
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r164082 | tilghman | 2008-12-13 17:22:02 -0600 (Sat, 13 Dec 2008) | 9 lines
Change the default calldurationlimit from the special value 0 to -1, so we
can better detect an exceptional case. This follows on to the changes made
in revision 156386. Related to issue #13851.
(closes issue #13974)
Reported by: paradise
Patches:
20081208__bug13974.diff.txt uploaded by Corydon76 (license 14)
Tested by: file, blitzrage, ZX81
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Sean Bright [Sat, 13 Dec 2008 18:25:58 +0000 (18:25 +0000)]
Use ast_str_strlen() instead of recalculating the string length.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164054
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Michiel van Baak [Sat, 13 Dec 2008 13:26:13 +0000 (13:26 +0000)]
nuke another use of the ast_str internals.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164028
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Tilghman Lesher [Sat, 13 Dec 2008 08:36:35 +0000 (08:36 +0000)]
Merge ast_str_opaque branch (discontinue usage of ast_str internals)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991
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Sean Bright [Sat, 13 Dec 2008 03:03:15 +0000 (03:03 +0000)]
This shouldn't have gotten commited. We might want to generate this into a separate file instead of the version controlled one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163952
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Sean Bright [Sat, 13 Dec 2008 03:00:26 +0000 (03:00 +0000)]
Use actual tables instead of ASCII art ones.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163951
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Joshua Colp [Sat, 13 Dec 2008 00:59:24 +0000 (00:59 +0000)]
Only detach and destroy the whisper audiohooks if they are actually in use.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163912
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Terry Wilson [Fri, 12 Dec 2008 23:48:26 +0000 (23:48 +0000)]
When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered.
(closes issue #14034)
Reported by: cristiandimache
Tested by: otherwiseguy, cristiandimache
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163873
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Russell Bryant [Fri, 12 Dec 2008 23:06:55 +0000 (23:06 +0000)]
Add a note to indicate why this only supports one channel for now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163828
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Russell Bryant [Fri, 12 Dec 2008 22:23:18 +0000 (22:23 +0000)]
Blocked revisions 163785 via svnmerge
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r163785 | russell | 2008-12-12 16:20:26 -0600 (Fri, 12 Dec 2008) | 2 lines
Set the reviewboard:url property on 1.4, as well
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Tilghman Lesher [Fri, 12 Dec 2008 22:04:26 +0000 (22:04 +0000)]
Merged revisions 163761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines
Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
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Russell Bryant [Fri, 12 Dec 2008 20:12:23 +0000 (20:12 +0000)]
Add a new CLI command, "channel redirect", which is similar in operation
to AMI Redirect.
Review: http://reviewboard.digium.com/r/89/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163716
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Steve Murphy [Fri, 12 Dec 2008 19:16:32 +0000 (19:16 +0000)]
demote always-appearing debug message (for certain boards) to ast_debug lev 3 msg instead
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163675
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Russell Bryant [Fri, 12 Dec 2008 18:45:03 +0000 (18:45 +0000)]
Rename a number of tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a relic from when
the struct was a server_instance, not a session_instance. It was renamed since
it represents both a server or client connection.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163670
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Russell Bryant [Fri, 12 Dec 2008 18:33:27 +0000 (18:33 +0000)]
Fix a small race condition in sip_tcp_locate().
We must increase the reference count on the tcptls_session _before_ unlocking
the thread list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163667
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Russell Bryant [Fri, 12 Dec 2008 18:19:47 +0000 (18:19 +0000)]
Resolve crashes when using SIP TCP/TLS with qualify.
The problem was a reference count error on the tcptls_session structure.
(closes issue #13989)
Reported by: Nugget
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163642
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Joshua Colp [Fri, 12 Dec 2008 18:17:12 +0000 (18:17 +0000)]
When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
(closes issue #13811)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163629
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Michiel van Baak [Fri, 12 Dec 2008 17:22:47 +0000 (17:22 +0000)]
Document default Monitor file location.
(closes issue #14065)
Reported by: kshumard
Patches:
res_monitor.documentation.patch.txt uploaded by kshumard (license 92)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163612
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Michiel van Baak [Fri, 12 Dec 2008 17:14:13 +0000 (17:14 +0000)]
Fix codec capability setup in chan_skinny
Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created.
Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order).
(closes issue #13806)
Reported by: pj
Patches:
codecs.diff uploaded by wedhorn (license 30)
Tested by: pj and me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163582
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Joshua Colp [Fri, 12 Dec 2008 16:55:15 +0000 (16:55 +0000)]
Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
(closes issue #13525)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163579
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Doug Bailey [Fri, 12 Dec 2008 15:10:25 +0000 (15:10 +0000)]
Add internationalization to sample configuration file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163516
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Russell Bryant [Fri, 12 Dec 2008 14:44:06 +0000 (14:44 +0000)]
Merged revisions 163511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) | 5 lines
Specify uint32_t for variables storing a CRC32 so that it is actually 32 bits
on 64-bit machines, as well.
(inspired by issue #13879)
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Russell Bryant [Fri, 12 Dec 2008 13:55:30 +0000 (13:55 +0000)]
Merged revisions 163448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
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Tilghman Lesher [Thu, 11 Dec 2008 23:38:56 +0000 (23:38 +0000)]
Merged revisions 163383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines
When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163384
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Matthew Nicholson [Thu, 11 Dec 2008 22:49:59 +0000 (22:49 +0000)]
Merged revisions 163316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec 2008) | 9 lines
Clean up the dundi cache every 5 minutes.
(closes issue #13819)
Reported by: adomjan
Patches:
pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
dundi_clearecache3.diff uploaded by mnicholson (license 96)
Tested by: adomjan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163317
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Russell Bryant [Thu, 11 Dec 2008 21:48:08 +0000 (21:48 +0000)]
Merged revisions 163253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines
Fix some observed slowdowns in dialplan processing.
The change is to remove autoservice usage from dialplan functions that do not
need it because they do not perform operations that potentially block.
(closes issue #13940)
Reported by: tbelder
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163254
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Russell Bryant [Thu, 11 Dec 2008 21:21:31 +0000 (21:21 +0000)]
Fix a problem where continuous mode will get inadvertently get turned off if set_rate()
is used while continuous mode was already turned on.
(closes issue #13738)
Reported by: smurfix
Patches:
res.patch.fixed uploaded by smurfix (license 547)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163241
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Mark Michelson [Thu, 11 Dec 2008 20:57:44 +0000 (20:57 +0000)]
Add an option to voicemail.conf to allow urgent messages to be
forwarded as not urgent.
(closes issue #14063)
Reported by: jaroth
Patches:
urgfwd_v2.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163213
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Mark Michelson [Thu, 11 Dec 2008 20:21:44 +0000 (20:21 +0000)]
Add an appropriate goto if ast_call fails
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163198
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Russell Bryant [Thu, 11 Dec 2008 20:07:47 +0000 (20:07 +0000)]
Fix the "failed" extension for outgoing calls.
The conversion to use ast_check_hangup() everywhere instead of checking the softhangup
flag directly introduced this problem. The issue is that ast_check_hangup() checked
for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances,
such as with a dummy channel.
The fix is simple. Don't check tech_pvt. It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets called. This happens inside
of ast_hangup(), which is the same function responsible for freeing the channel. Any
code calling ast_check_hangup() better not be calling it after that point, and if so,
we have a bigger problem at hand.
(closes issue #14035)
Reported by: erogoza
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163171
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Tilghman Lesher [Thu, 11 Dec 2008 20:02:35 +0000 (20:02 +0000)]
Sometimes even Linux needs -lm to link libtonezone, such as when libtonezone
is compiled statically.
(closes issue #13887)
Reported by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163168
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Mark Michelson [Thu, 11 Dec 2008 19:40:18 +0000 (19:40 +0000)]
Reduce indentation level of ast_feature_request_and_dial
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163166
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Russell Bryant [Thu, 11 Dec 2008 17:06:16 +0000 (17:06 +0000)]
Merged revisions 163092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines
Fix an issue that made it so you could only have a single caller executing
a custom feature at a time. This was especially problematic when custom
features ran for any appreciable amount of time.
The fix turned out to be quite simple. The dynamic features are now stored
in a read/write list instead of a list using a mutex.
(closes issue #13478)
Reported by: neutrino88
Fix suggested by file
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Tilghman Lesher [Thu, 11 Dec 2008 16:52:24 +0000 (16:52 +0000)]
Merged revisions 163088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) | 6 lines
Don't wait forever, if there's a specified recording timeout.
(closes issue #13885)
Reported by: bamby
Patches:
res_agi.c.patch uploaded by bamby (license 430)
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Mark Michelson [Thu, 11 Dec 2008 16:47:34 +0000 (16:47 +0000)]
Merged revisions 163084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines
Revert this cast to long. Using time_t here causes build failures on a
FreeBSD 32-bit build.
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Mark Michelson [Thu, 11 Dec 2008 16:33:16 +0000 (16:33 +0000)]
Merged revisions 163080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines
Fix a potential crash due to unsafe datastore handling.
This patch also contains a conversion from using long to time_t
for representing times for a queue, as well as some whitespace
fixes.
(closes issue #14060)
Reported by: nivek
Patches:
datastore_fixup.patch.corrected uploaded by nivek (license 636)
with slight modification from me
Tested by: nivek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163081
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Sean Bright [Thu, 11 Dec 2008 15:40:07 +0000 (15:40 +0000)]
Fix some of the grammar issues in doc/tex/qos.tex.
(closes issue #14049)
Reported by: kshumard
Patches:
doc.tex.qos.tex.patch uploaded by kshumard (license 92)
(Slight modifications by seanbright)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163037
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Joshua Colp [Thu, 11 Dec 2008 15:05:49 +0000 (15:05 +0000)]
When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it.
(closes issue #13525)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162997
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Tilghman Lesher [Wed, 10 Dec 2008 23:01:14 +0000 (23:01 +0000)]
Previously missing line, now the substitution works correctly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162930
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Jeff Peeler [Wed, 10 Dec 2008 22:53:34 +0000 (22:53 +0000)]
Merged revisions 162926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) | 3 lines
Oops, inverted logic for a strcasecmp check. Pointed out by mmichelson, thanks!
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Joshua Colp [Wed, 10 Dec 2008 22:48:58 +0000 (22:48 +0000)]
Fix reloads of aliased CLI commands. Due to changes done to turn it into a single memory allocation we can't just use the existing CLI alias structure. We have to destroy all existing ones and then create new ones.
(closes issue #14054)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162923
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Tilghman Lesher [Wed, 10 Dec 2008 22:48:09 +0000 (22:48 +0000)]
Checking global variables here actually overwrote the previous substitution by
channel variables, and in any case, was redundant;
pbx_substitute_variables_helper ALREADY does substitution for global
variables.
(closes issue #13327)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162922
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Jeff Peeler [Wed, 10 Dec 2008 22:11:46 +0000 (22:11 +0000)]
Merged revisions 162874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines
(closes issue #13229)
Reported by: clegall_proformatique
Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams.
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Joshua Colp [Wed, 10 Dec 2008 19:02:57 +0000 (19:02 +0000)]
Merged revisions 162804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines
Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
(closes issue #12560)
Reported by: vsauer
Patches:
patch001.diff uploaded by ramonpeek (license 266)
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Joshua Colp [Wed, 10 Dec 2008 17:53:09 +0000 (17:53 +0000)]
Merged revisions 162738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines
When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599)
Reported by: hjourdain
Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
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Michiel van Baak [Wed, 10 Dec 2008 17:09:15 +0000 (17:09 +0000)]
add tab completion for 'core set debug X filename.c'
(closes issue #13969)
Reported by: jtodd
Patches:
20081205__bug13969.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak, eliel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162687
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Steve Murphy [Wed, 10 Dec 2008 16:52:05 +0000 (16:52 +0000)]
Blocked revisions 162671 via svnmerge
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r162671 | murf | 2008-12-10 09:45:01 -0700 (Wed, 10 Dec 2008) | 22 lines
(closes issue #14022)
Reported by: wetwired
Tested by: murf
I checked, and I added a mod to the trunk version
of Asterisk that would make it 8-bit transparent
on 27 Nov 2007, but I made no such updates to
1.4. My best guess is that 1.4 was released, and
it was not appropriate to commit an enhancement.
But I'm going to add the same fixes to 1.4 now,
for the following reasons:
1. wetwired is correct; 1.4 is **mostly** 8-bit
transparent now. This is because the lexical
token forming rules use . in most 'word'
state continuances. It's just the beginning
of a 'word' that is picky.
2. Accepting 8-bit chars in some places and
not others leads to bug reports like this.
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Mark Michelson [Wed, 10 Dec 2008 16:46:51 +0000 (16:46 +0000)]
Blocked revisions 162670 via svnmerge
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r162670 | mmichelson | 2008-12-10 10:44:37 -0600 (Wed, 10 Dec 2008) | 14 lines
Update to stringfield handling so that side-effects on
parameters are not evaluated multiple times.
An example where this caused a problem was in chan_sip.c, with
the line
ast_string_field_set(p, fromdomain, ++fromdomain);
This patch was originally uploaded to issue #13783 by
jamessan. While the issue was closed for other reasons, this
patch is valid and fixes a separate problem, and is thus
being committed.
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Mark Michelson [Wed, 10 Dec 2008 16:39:10 +0000 (16:39 +0000)]
Merged revisions 162659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines
Add missing documentation to misdn.txt
(closes issue #14052)
Reported by: festr
Patches:
misdn.txt.patch uploaded by festr (license 443)
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