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r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines
Merged revisions 170147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282)
Reported by: cheesegrits
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r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines
Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop.
(closes issue #14304)
Reported by: jcovert
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r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines
Fix device state parsing issues for channel names with multiple slashes
The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.
In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.
(closes issue #14014)
Reported by: kebl0155
Patches:
14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155
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r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines
Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
(closes issue #14117)
Reported by: sergedevorop
Patches:
20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
Tested by: sergedevorop
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r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines
Merged revisions 168628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
(closes issue #14086)
Reported by: ZX81
Patches:
14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr
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r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines
Merged revisions 168593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128.
Proper bounds checking was not in place to enforce this limit, so if more than
128 extensions were passed to the Page() app, Asterisk would crash. This patch
instead dynamically allocates memory for the ast_dial structures and removes
the (non-functional) arbitrary limit.
This issue would have special importance to anyone who is dynamically creating
the argument passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external interface.
The patch posted by a_villacis was slightly modified for some coding guidelines
and other cleanups. Thanks, a_villacis!
(closes issue #14217)
Reported by: a_villacis
Patches:
20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
Tested by: otherwiseguy
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r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines
Update app_queue to deal with the removal of AST_PBX_KEEPALIVE
When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.
I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.
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Due to non-symmetrical updating, I had some fairly
interesting conflicts to straighten out in this
release. The changes were such that I was compelled
to run thru all the same tests as trunk, which turned
up some problems, which I fixed.
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r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
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r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
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r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines
Merged revisions 165889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines
Ensure that the chanspy datastore is fully initialized.
This patch resolved some random crash issues observed by a user on a BSD system
(closes issue #14111)
Reported by: ys
Patches:
app_chanspy.c.diff uploaded by ys (license 281)
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r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines
Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.
This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source. While this usage was perfectly safe,
there are others that are problematic. Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.
Further changes to get rid of KEEPALIVE and related code is being done by
murf. There is a patch up for that on review 29.
Review: http://reviewboard.digium.com/r/98/
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r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines
Merged revisions 165255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines
Fix some memory leaks found while looking at how realtime
configs are handled.
Also cleaned up some coding guidelines violations in app_realtime.c,
mostly related to spacing
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r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, 17 Dec 2008) | 10 lines
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.
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r165143 | mmichelson | 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines
And actually assign the function to a pointer...
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r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines
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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r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines
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
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r163873 | twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines
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
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r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec 2008) | 22 lines
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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r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines
Merged revisions 162286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
(closes issue #12471)
Reported by: mthomasslo
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r161493 | mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 lines
If the autoloop flag is set on a channel, then we need to
add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.
This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.
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r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, 05 Dec 2008) | 5 lines
When using IMAP_STORAGE, it's important to convert bare newlines (\n) in
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed
by Mark M. on IRC.
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r161350 | seanbright | 2008-12-05 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines
Use ast_free() instead of free(), pointed out by eliel on IRC.
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r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines
Add some safety measures when using gosub, especially when using the options
for app_dial and app_queue to run a gosub when the call is answered.
* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
then this will cause errors when we attempt to actually run the gosub, including
a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
to actually run the gosub routine. If there was an error, we should not attempt
to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.
(closes issue #13548)
Reported by: fiddur
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r160555 | mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 lines
When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.
With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.
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r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) | 16 lines
Merged revisions 156178 via svnmerge from
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r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines
(closes issue #13173)
Reported by: pep
This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference.
Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/
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r152969 | tilghman | 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) | 10 lines
Merged revisions 152958 via svnmerge from
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r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines
Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400)
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r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines
Merged revisions 153114 via svnmerge from
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r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines
Turn off qualify on uncached realtime peers.
(Closes issue #13383)
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r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines
Recorded merge of revisions 154263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines
Make the monitor thread non-detached, so it can be joined (suggested by Russell
on -dev list).
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r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines
Merged revisions 154266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines
JIRA ABE-1703
mISDN sets the channel to the wrong state when it receives
the indication AST_CONTROL_RINGING.
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r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines
Merged revisions 154365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines
On busy systems, it's possible for the values checked within a single line
of code to change, unless the structure is locked to ensure a consistent
state.
(closes issue #13717)
Reported by: kowalma
Patches:
20081102__bug13717.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
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r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines
Merged revisions 155398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines
Clarify error message.
(closes issue #13809)
Reported by: denke
Patches:
20081104__bug13809.diff.txt uploaded by Corydon76 (license 14)
Tested by: denke
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r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines
Merged revisions 155861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines
Channel drivers assume that when their indicate callback
is invoked, that the channel on which the callback was called
is locked. This patch corrects an instance in chan_agent where
a channel's indicate callback is called directly without first
locking the channel.
This was leading to some observed locking issues in chan_local,
but considering that all channel drivers operate under the
same expectations, the generic fix in chan_agent is the right
way to go.
AST-126
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r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines
Merged revisions 156164 via svnmerge from
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r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines
Move the sanity check that makes sure "always fork" is not set along with the
console option to be after the code that reads options from asterisk.conf.
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.)
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r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines
Merged revisions 156294 via svnmerge from
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r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines
If the SLA thread is not started, then reload causes a memory leak.
(closes issue #13889)
Reported by: eliel
Patches:
app_meetme.c.patch uploaded by eliel (license 64)
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r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines
Merged revisions 156688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines
Provide more space for all the data which can appear in an originating
channel name.
(closes issue #13398)
Reported by: bamby
Patches:
manager.c.diff uploaded by bamby (license 430)
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r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines
Merged revisions 156755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
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r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
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r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
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r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines
Merged revisions 158539 via svnmerge from
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r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines
When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock
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r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines
Merged revisions 158600 via svnmerge from
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r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines
The passed extension may not be the same in the list as the current entry,
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
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r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines
Merged revisions 159269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines
Don't try to send a response on a NULL pvt.
(closes issue #13919)
Reported by: barthpbx
Patches:
chan_iax2.c.patch uploaded by eliel (license 64)
Tested by: barthpbx
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
