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r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines
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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines
Tested by: sergee, murf, chris-mac, andrew, KNK
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.
** trunk note: some code to suppress the h exten being run
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.
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r142576 | murf | 2008-09-11 17:12:53 -0600 (Thu, 11 Sep 2008) | 28 lines
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r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | 20 lines
(closes issue #13364)
Reported by: mdu113
Well, fundamentally, the problems revealed in 13364 are
because of the ForkCDR call that is done before the dial.
When the bridge is in place, it's dealing with the first
(and wrong) cdr in the list.
So, I wrote a little func to zip down to the first non-locked
cdr in the chain, and thru-out the ast_bridge_call, these
results are used instead of raw chan->cdr and peer->cdr pointers.
This shouldn't affect anyone who isn't forking cdrs before a
dial, and should correct the cdr's of those that do.
So, this change ends up correcting the dstchannel
and userfield; the disposition was fixed by a previous
patch, it was OK coming into this problem.
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r142475 | murf | 2008-09-10 16:11:27 -0600 (Wed, 10 Sep 2008) | 38 lines
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r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | 30 lines
(closes issue #12318)
Reported by: krtorio
I made a small change to the code that handles local channel situations.
In that code, I copy the answer time from the peer cdr, to the bridge_cdr,
but I wasn't also copying the disposition from the peer cdr.
So, Now I copy the disposition, and I've tested against
these cases:
1. phone 1 never answers the phone; no cdr is generated at all.
this should show up as a manager command failure or something.
2. phone 2 never answers. CDR is generated, says NO ANSWER
3. phone 2 is busy. CDR is generated, says BUSY
4. phone 2 answers: CDR is generated, times are correct; disposition
is ANSWERED, which is correct. The start time is the time that
the manager dialed the first phone. The answer time is the time
the second phone picks up.
I purposely left the cid and src fields blank; since this call really
originates from the manager, there is no 'easy' data to put in these
fields. If you feel strongly that these fields should be filled in,
re-open this bug and I'll dig further.
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r142355 | russell | 2008-09-10 11:41:55 -0500 (Wed, 10 Sep 2008) | 15 lines
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r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) | 7 lines
It is a normal situation that a task gets put in the scheduler that should run
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)
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r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines
Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep()
or when calling ast_waitfor(). These are inappropriate times to hold the channel
lock. This is what has caused "could not get the channel lock" messages from
chan_sip and has likely caused a negative impact on performance results of SIP
in Asterisk 1.6. Thanks to file for pointing out this section of code.
(closes issue #13287)
(closes issue #13115)
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r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) | 15 lines
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r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines
When doing an async goto, detect if the channel is already in the middle of a
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
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r141039 | jpeeler | 2008-09-04 12:27:56 -0500 (Thu, 04 Sep 2008) | 15 lines
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r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines
(closes issue #11979)
Fixes multiple parking problems:
Crash when executing a park on an extension dialed by AGI due to not returning the proper return code.
Crash when using a builtin feature that was a subset of a enabled dynamic feature.
Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked.
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r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | 11 lines
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r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line
I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug.
For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them.
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r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines
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r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line
After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints.
Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations,
where you'd want to post single-channel cdrs.
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r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | 14 lines
Merged revisions 139635 via svnmerge from
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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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r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines
Merged revisions 139347 via svnmerge from
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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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r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | 20 lines
Merged revisions 139074 via svnmerge from
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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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r138815 | murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines
These changes are in regards to bug 13249, where users are being surprised by the changes made
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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r138482 | seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 lines
Move Uniqueid to the end of the event for those that rely on the position
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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r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) | 17 lines
Merged revisions 138027 via svnmerge from
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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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r136633 | mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 lines
Fix a calculation error I had made in the poll. The poll
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
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r136631 | mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 lines
Scrap the 500 ms delay when Asterisk auto-answers a channel.
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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r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines
Merged revisions 136062 via svnmerge from
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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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r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines
Merged revisions 135841,135847,135850 via svnmerge from
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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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r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines
Merged revisions 135799 via svnmerge from
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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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r135265 | murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines
(closes issue #13202)
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.
in 1.6.0, the changes to only main/pbx.c were applicable,
as apparently the code added to main/features by jpeeler
were not included in 1.6.0.
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r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | 63 lines
Merged revisions 134883 via svnmerge from
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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.
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r134125 | mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 lines
This commit compensates for buggy poll(2)
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)
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r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 Jul 2008) | 6 lines
ast_device_state() gets called in two different ways. The first way is when
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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r133946 | russell | 2008-07-26 10:16:20 -0500 (Sat, 26 Jul 2008) | 1 line
actually use the cache_cache argument
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r133299 | murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines
(closes issue #13144)
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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