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r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines
(closes issue #13409)
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.
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r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
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.
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Reported by: erousseau
This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it
could only be applied to trunk.
Basically, for accounting where "initiated" seconds
are billed for, if the microseconds field on the end
time is greater than the microseconds field for the
answer time, add one second to the billsec field.
The implementation was requested by erousseau, and
I've implemented it as requested. I've updated the
CHANGES, the cdr.conf.sample, and the .h files
accordingly, to accept and set a flag for the
corresponding new option. cdr.c adds in the extra
second based on the usec fields if the option is
set. Tested, seems to be working fine.
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r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines
This patch reverts the changes made via 139347, and 139635, as users
are seeing adverse difference.
I will un-close 13251.
Back to the drawing board/ concept/ beginning/ whatever!
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r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines
I found some problems with the code I committed earlier, when
I merged them into trunk, so I'm coming back to clean up.
And, in the process, I found an error in the code I added
to trunk and 1.6.x, that I'll fix using this patch also.
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r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines
(closes issue #13251)
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!
........
I also made a little fix to the app_dial's 'e' option,
that is related to my updates.
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r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines
(closes issue #13263)
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.
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to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if
they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x
installation where a "make samples" was executed, or where they hand-edited the
asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher).
(this commit does not totally solve 13249, at least not yet)
The change involves issueing a single warning while the AEL file is loading, if:
1. app_set is present in the config file, and set to 1.6 or higher.
2. there are double quotes in an assignment statement (eg x = "hi there";)
3. the warning was not already issued.
The standalone app, aelparse, does not (yet) issue this warning. I'd have to
have it read in the asterisk.conf file, and that's a bit of hassle. I'll add
it if users request it, tho.
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of the name/value pairs, pointed out by snuffy-home on #asterisk-commits.
For those of you who rely on the position of name/value pairs in manager
events... stop... that is why associative arrays were invented.
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r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines
Ensure that when a hangup occurs in autoservice, that a hangup frame gets
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)
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would reset to 500 ms every time a non-voice frame
was received. The total time we poll should be 500 ms, so
now we save the amount of time left after the poll returned
and use that as our argument for the next call to poll
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Instead, poll the channel until receiving a voice frame. The
cap on this poll is 500 ms.
The optional delay is still allowable in the Answer() application,
but the delay has been moved back to its original position, after
the call to the channel's answer callback. The poll for the voice
frame will not happen if a delay is specified when calling Answer().
(closes issue #12708)
Reported by: kactus
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r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines
Since adding the AST_CONTROL_SRCUPDATE frame type,
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
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r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines
Merging the issue11259 branch.
The purpose of this branch was to take into account
"burps" which could cause jitterbuffers to misbehave.
One such example is if the L option to Dial() were used
to inject audio into a bridged conversation at regular
intervals. Since the audio here was not passed through
the jitterbuffer, it would cause a gap in the jitterbuffer's
timestamps which would cause a frames to be dropped for a
brief period.
Now ast_generic_bridge will empty and reset the jitterbuffer
each time it is called. This causes injected audio to be handled
properly.
ast_generic_bridge also will empty and reset the jitterbuffer
if it receives an AST_CONTROL_SRCUPDATE frame since the change
in audio source could negatively affect the jitterbuffer.
All of this was made possible by adding a new public API call
to the abstract_jb called ast_jb_empty_and_reset.
(closes issue #11259)
Reported by: plack
Tested by: putnopvut
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r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines
Revert inadvertent changes to app_skel that occurred when
I was testing for a memory leak
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r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines
Remove properties that should not be here
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r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
........
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Reported by: falves11
Tested by: murf
falves11 ==
The changes I introduce here seem to clear up the problem
for me. However, if they do not for you, please reopen this
bug, and we'll keep digging.
The root of this problem seems to be a subtle memory corruption
introduced when creating an extension with an empty extension
name. While valgrind cannot detect it outside of DEBUG_MALLOC
mode, when compiled with DEBUG_MALLOC, this is certain death.
The code in main/features.c is a puzzle to me. On the initial
module load, the code is attempting to add the parking extension
before the features.conf file has even been opened!
I just wrapped the offending call with an if() that will not
try to add the extension if the extension name is empty. THis
seems to solve the corruption, and let the "memory show allocations"
work as one would expect.
But, really, adding an extension with an empty name is a seriously
bad thing to allow, as it will mess up all the pattern matching
algorithms, etc. So, I added a statement to the add_extension2 code to return
a -1 if this is attempted.
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r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines
(closes issue #11849)
Reported by: greyvoip
Tested by: murf
OK, a few days of debugging, a bunch of instrumentation
in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid
notebook pages of notes later, I have made the small
tweek necc. to get the start time right on the second
CDR when:
A Calls B
B answ.
A hits Xfer button on sip phone,
A dials C and hits the OK button,
A hangs up
C answers ringing phone
B and C converse
B and/or C hangs up
But does not harm the scenario where:
A Calls B
B answ.
B hits xfer button on sip phone,
B dials C and hits the OK button,
B hangs up
C answers ringing phone
A and C converse
A and/or C hangs up
The difference in start times on the second CDR is because
of a Masquerade on the B channel when the xfer number is
sent. It ends up replacing the CDR on the B channel with
a duplicate, which ends up getting tossed out. We keep
a pointer to the first CDR, and update *that* after the
bridge closes. But, only if the CDR has changed.
I hope this change is specific enough not to muck
up any current CDR-based apps. In my defence, I
assert that the previous information was wrong,
and this change fixes it, and possibly other
similar scenarios.
I wonder if I should be doing the same thing
for the channel, as I did for the peer, but
I can't think of a scenario this might affect.
I leave it, then, as an exersize for the users,
to find the scenario where the chan's CDR
changes and loses the proper start time.
........
and as to 1.4 to trunk; have I expressed my
feelings about code shifting from one file
to another? Good.
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driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
respectively. Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
Reported by: eliel
Patches:
cliprompt.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.
On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.
Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.
closes issue #11928)
Reported by: adriavidal
Patches:
1.6.0-configurev2.patch uploaded by putnopvut (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
called from elsewhere in Asterisk to find the current state of a device. In
that case, we want to use the cached value if it exists. The other way is when
processing a device state change. In that case, we do not want to check the
cache because returning the last known state is counter productive.
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we do NOT need to uri_decode in manager.
(if I sent core%20show%20channels from a telnet
session, it should be interpreted literally, however,
if I send that from an http session, it should be
decoded, which is the behaivor now)
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r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines
Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
Reported by: davidw
Patches:
20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw, jvandal, murf
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ago (does not affect 1.4), where you would pass
a pointer to the end of a character array, and
ast_uri_decode would do no good.
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to Asterisk licensing information. The licensing page includes the Asterisk license,
as well as a (not yet complete) list of 3rd party libraries that may be used, as well
as what license we receive them under.
Help filling out this list in the format that I have started in doxyref.h would be
much appreciated. :)
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the code checks to see if there is a cached state available, use the aggregate
cached state across all servers, and not just the local state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
this commit, only the logger thread's PID would
be printed.
(closes issue #13150)
Reported by: atis
Patches:
log_pid.diff uploaded by putnopvut (license 60)
Tested by: eliel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: murf
Tested by: murf
For: J. Geis
The 'data' field in the ast_exten struct was being
'moved' from the current dialplan to the replacement
dialplan. This was not good, as the current dialplan
could have problems in the time between the change
and when the new dialplan is swapped in.
So, I modified the merge_and_delete code to strdup
the 'data' field (the args to the app call), and
then it's freed as normal.
I improved a few messages; I added code to limit
the number of calls to the context_merge_incls_swits_igps_other_registrars()
to one per context. I don't think having it called
multiple times per context was doing anything bad,
but it was inefficient.
I hope this fixes the problems Mr. Geiss was noting in
asterisk-users, see
http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html
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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
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r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines
minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool)
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own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
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removed early (before the routine to load the configuration was
finished) because a variable wasn't initialized.
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probably not a good idea, as we might run out of stack space. Therefore,
changing this over to use the ast_str infrastructure for buffers is
probably a good idea.
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the same keyword as the other files (patch by eliel).
(closes issue #13104)
Reported by: eliel
Patches:
revision.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a feature without specifying a group or feature to register.
(closes issue #13101)
Reported by: eliel
Patches:
features.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
calculate the number of bytes from a sysinfo structure.
unsigned long.
(closes issue #13057)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) | 10 lines
notify the user that dnsmgr refresh wont work when dnsmgr is not enabled.
Previously this command would automagically appear and disappear.
This was confusing.
(closes issue #12796)
Reported by: chappell
Patches:
dnsmgr_refresh_3.diff uploaded by chappell (license 8)
Tested by: russell, chappell, mvanbaak
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: eliel
OK, now the context registrar slot is strdup'd. It is freed
on destruction. I don't see the need to do this with all
the structs' registrar fields, but if some wild case proves
they should also be handled this way, then we can
put in the extra work at that time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Jeff Peeler
Mark Michelson
Tilghman Lesher
Sean Bright
Russell Bryant
Dwayne M. Hubbard
Kevin P. Fleming
Terry Wilson
Brett Bryant
Brandon Kruse
Pari Nannapaneni
Michiel van Baak