channel list calling a caller-defined callback. The callback returns non-zero
if a match is found. This should speed up some of the code that I committed
earlier today in chan_sip (which is also updated by this commit).
Reviewed by russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
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had no effect
* Updated dialing API documentation to indicate that timeouts
are specified in milliseconds
* Added a new timeout argument to the Page application. If time
expires, any endpoints which have not answered will be hung up.
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_must_ be increased before creating the scheduler entry. Otherwise, you
create a race condition where the reference count may hit zero and the
object can disappear out from under you. This could also would have
incorrectly decreased the reference count in the case that the scheduler
add failed.
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r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
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.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
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.
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1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines
2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)
3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)
4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied
5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
Add a tolerance period for sync-triggered audiohooks
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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'update2', which permits updates which match across multiple columns, instead
of requiring all tables to have a single unique identifier. All of the other
API calls with the exception of 'update' already had the ability to match on
multiple fields, so it was a missing and very desireable feature that an API
call implementing an update should have this, too.
This does not change any outward performance of Asterisk, but it should make
life easier for application developers who use the RealTime framework.
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app_voicemail. Instead, include it where it is needed. This turned out to be a
relatively minor issue because other headers include logger.h as well.
Need to test -addons before merging this back to 1.6.0.
(closes issue #13605)
Reported by: tomo1657
Patches:
13605_seanbright.diff uploaded by seanbright (license 71)
Tested by: mmichelson
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Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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This allows for the ODBC parts to work on OpenBSD as well.
99.99% of the work is done by seanbright (bow, bow) and I actually
did nothing but test and yell at him that it still didn't work :)
Thanks for helping out !
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problem within strptime(3), which we are correcting here with ast_strptime().
(closes issue #11040)
Reported by: DEA
Patches:
20080910__bug11040.diff.txt uploaded by Corydon76 (license 14)
Tested by: DEA
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r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines
improve header inclusion process in a few small ways:
- 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
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r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines
fix some minor issues with rev 144924
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It was pretty sparsely documented.
This update fleshes out the pbx_lua module, to
add the switch statements to the extensions in the
extensions.lua file, as well as removing them when
the module is unloaded.
Many thanks to Matt Nicholson for his fine
contribution!
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when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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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r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines
(closes issue #13236)
Reported by: korihor
Wow, this one was a challenge!
I regrouped and ran a new strategy for
setting the ~~MACRO~~ value; I set it once
per extension, up near the top. It is only
set if there is a switch in the extension.
So, I had to put in a chunk of code to detect
a switch in the pval tree.
I moved the code to insert the set of ~~exten~~
up to the beginning of the gen_prios routine,
instead of down in the switch code.
I learned that I have to push the detection
of the switches down into the code, so everywhere
I create a new exten in gen_prios, I make sure
to pass onto it the values of the mother_exten
first, and the exten next.
I had to add a couple fields to the exten
struct to accomplish this, in the ael_structs.h
file. The checked field makes it so we don't
repeat the switch search if it's been done.
I also updated the regressions.
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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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russell