since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@159316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
frame in a freed ast_filestream. This patch makes use of the
ao2 functions to make sure that we do not free an ast_filestream
structure until the embedded ast_frame has been "freed" as well.
(closes issue #13496)
Reported by: fst-onge
Patches:
filestream_frame_1_4.diff uploaded by putnopvut (license 60)
Tested by: putnopvut
Closes AST-89
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@158126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@157859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev; he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been.
This change, tested on these 3 patterns now
picks the proper one.
However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half
years or so now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156164 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
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
........
r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
........
r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
........
r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
........
r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@145293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose
- astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled
- simplify the usage of some of these headers in the AEL-related stuff in the utils directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
does not apply since the "found" pointer is not
passed in to this function. If this is going to
be backported, it needs to be done differently or
a deeper backport needs to be done.
Edit: This commit reverts commit number 144677.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: mnicholson
Patches:
found1.diff uploaded by mnicholson (license 96)
This patch was mainly meant to apply to trunk and 1.6.x,
but I'm applying it to 1.4 also, which should be a perfectly
harmless fix to the vast majority of users who are not using
external switches, but the few who might be affected
will not have to go to the pain of filing a bug report.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
as soon as possible. Accept "0" as an acceptable time to run, and also treat
negative as "run now", and don't print a debug message about it.
(inspired by a message asking about the "request to schedule in the past"
debug message on the -dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
masquerade. This can happen when chan_local is trying to optimize itself out.
If this happens, fail the async goto instead of bursting into flames.
(closes issue #13435)
Reported by: geoff2010
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@141806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: tomaso
Patches:
asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564)
I basically spent the day, verifying that this patch
solves the problem, and doesn't hurt in non-problem
cases. Why valgrind did not plainly reveal this leak
absolutely mystifies and stuns me.
Many, many thanks to tomaso for finding and providing the fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@140670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@140488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
UpdateConfig manager action is issued and the
file specified in DstFileName does not yet exist,
an error is not returned.
(closes issue #13341)
Reported by: vadim
Patches:
13341.patch uploaded by putnopvut (license 60)
(with small modification from seanbright)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@139769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: sergee
Tested by: murf
THis is a bold move for a static release fix, but I wouldn't have
made it if I didn't feel confident (at least a *bit* confident)
that it wouldn't mess everyone up.
The reasoning goes something like this:
1. We simply cannot do anything with CDR's at the current point
(in pbx.c, after the __ast_pbx_run loop). It's way too late to
have any affect on the CDRs. The CDR is already posted and gone,
and the remnants have been cleared.
2. I was very much afraid that moving the running of the 'h'
extension down into the bridge code (where it would be now
practical to do it), would result in a lot more calls to the
'h' exten, so I implemented it as another exten under another
name, but found, to my pleasant surprise, that there was a
1:1 correspondence to the running of the 'h' exten in the
pbx_run loop, and the new spot at the end of the bridge.
So, I ifdef'd out the current 'h' loop, and moved it into
the bridge code. The only difference I can see is the stuff
about the AST_PBX_KEEPALIVE, and hopefully, if this
is still an important decision point, I can replicate it
if there are complaints. To be perfectly honest,
the KEEPALIVE situation is not totally clear to me,
and how it relates to a post-bridge situation is less
clear. I suspect the users will point out everything
in total clarity if this steps on anyone's toes!
3. I temporarily swap the bridge_cdr into the channel
before running the 'h' exten, which makes it possible
for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set,
the users can also get at the billsec/duration vals.
After the h exten finishes, the cdr is swapped back
and processing continues as normal.
Please, all who deal with CDR's, please test this version
of Asterisk, and file bug reports as appropriate!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@139347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: brainy
Tested by: murf
The specialized reset routine is tromping on the
flags field of the CDR. I made a change to not
reset the DISABLED bit. This should get rid of this
problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@139074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
properly deferred to be read from the channel owner when it gets taken out
of autoservice.
(closes issue #12874)
Reported by: dimas
Patches:
v1-12874.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@138027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@136062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
