dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.
I hope this doesn't spoil some vast, eternal plan...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
per call (which could otherwise lead to inconsistent status reports).
(closes issue #13216)
Reported by: ruddy
Patches:
20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
Tested by: ruddy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
match the trailing \0 of a string) we need to check that first, otherwise
we end up with incorrect results. Fix suggested by reporter.
(closes issue #13787)
Reported by: meitinger
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
deadlock when someone tries to initiate a module reload from the AMI just
as Asterisk is starting.
(closes issue #13778)
Reported by: hotsblanc
Fix suggested by hotsblanc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem was that if someone parked a call, the callee and caller would both get assigned the builtin transfer feature, which would not only be potentially giving someone the ability to transfer themselves when they shouldn't have it, but would also dissallow reinviting the media off of the call.
(closes issue #12854)
Reported by: davidw
Patches:
parkingfix4.diff.txt uploaded by otherwiseguy
Tested by: davidw, otherwiseguy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A comment was made in bug #13726
"3. The same mistake as in (2) is done in a few other places in the code that check for: #if defined(HAVE_ZAPTEL) || defined(HAVE_DAHDI)
Harmless, but still incorrect."
In the case of main/asterisk.c, this is not incorrect because without HAVE_ZAPTEL defined, we're missing
the include for ioctl and the namespace that defines DAHDI_TIMERCONFIG which is still required when
using Zaptel with the 1.4 branch.
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We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing.
(closes issue #13715)
reported by: makoto
patch by: bweschke
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bridge with peer completely GONE.
I moved the channel find call up to cover the
whole peer CDR reset code segment. This appears
to solve the crash without changing the logic
at all.
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Don't always define HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..)
Minor cleanup to make things clear.
(closes issue #13726)
Reported by: tzafrir
Patches:
dahdi_def.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I was mistakenly under the assumption that dialplan functions
*always* required that a channel be present. I need to go
home earlier, I think :)
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the manager's GetVar handler if an invalid channel has
been specified. Several dialplan functions, including
CHANNEL and SIP_HEADER, do not check for NULL-ness of
the channel being passed in.
(closes issue #13715)
Reported by: makoto
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ABE-1628 and RYM-150398 and AST-103 in internal Digium
bug trackers.
These fixes address a really subtle memory corruption
problem that would happen in machines heavily loaded
in production environments. The corruption would
always take the form of the STMT object getting
nulled out and one of the unixODBC calls would
crash trying to access statement->connection.
It isn't fully proven yet, but the server has
now been running 2.5 days without appreciable
memory growth, or any gain of %cpu, and no
crashes. Whether this is the problem or not
on that server, these fixes are still warranted.
As it turns out, **I** introduced these errors
unwittingly, when I corrected another crash earlier.
I had formed the build_query routine, and failed
to remove mutex_unlock calls in 3 places in the
transplanted code. These unlocks would only
happen in error situations, but unlocking the
mutex early set the code up for a catastrophic
failure, it appears. It would happen only once
every 100K-200K or more calls, under heavy load...
but that is enough.
If another crash occurs, with the same MO,
I'll come back and remove my confession from the log, and
we'll keep searching, but the fact that we
have Asterisk dying from an asynchronous
wiping of the STMT object, only on some connection
error, and that the server has lived for 2.5
days on this code without a crash, sure make
it look like this was the problem!
Also, in several points, Statement handles are
set to NULL after SQLFreeHandle. This was mainly
for insurance, to guarantee a crash. As it turns
out, the code does not appear to be attempting
to use these freed pointers.
Asterisk owes a debt of gratitude to Federico Alves
and Frediano Ziglio for their untiring efforts in
finding this bug, among others.
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so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
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whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.
(closes issue #13395)
Reported by: Marquis
Patches:
app_queue.c.transfer.patch uploaded by Marquis (license 32)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: krisk84
Tested by: krisk84
This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@147941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@146799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
method, because we need to evaluate the callerid (since that data affects
whether an extension exists).
(closes issue #13343)
Reported by: efutch
Patches:
20080915__bug13343.diff.txt uploaded by Corydon76 (license 14)
Tested by: efutch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@146711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
