https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: murf
Most of this bug was already fixed by Tilghman before
I opened it; Many thanks to Tilghman for his fix
in svn version 125794. That fix cleared up some of the
fields in the lock_info.
This commit changes the address that is stored for the
lock in the lock_info struct, so that it is the same
as that passed into the locking macros. This makes
searching for a lock_info (as in log_show_lock())
by its lock addr possible. The lock_addr field is
infinitely more useful if it is the same as what
is 'publicly' available outside the lock_info code.
Many thanks to kpfleming, putnopvut, and Russell for their
invaluable insights earlier today.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines
Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch,
and b) completes contexts correctly when the extension is ambiguous.
(closes issue #12980)
Reported by: licedey
Patches:
20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to document arguments seems logical, and does work, but is not the best
thing to use.
\arg in doxygen is simply for creating non-nested unordered lists. \param is
the correct tag to use to document function parameters, and will come out
better in the generated documentation.
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r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) | 10 lines
Fix a typo in the non-DEBUG_THREADS version of the recently added DEADLOCK_AVOIDANCE()
macro. This caused the lock to not actually be released, and as a result, not
avoid deadlocks at all. This resolves the issues reported in the last while about
Asterisk locking up all over the place (and most commonly, in chan_iax2).
(closes issue #12927)
(closes issue #12940)
(closes issue #12925)
(potentially closes others ...)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
implementation from using the volatile libtds, to using the db-lib front end.
The unintended side effect of this is that we support (at least) versions 0.62
through 0.82 of the FreeTDS distribution without any #ifdef ugliness.
(closes issue #12844)
Reported by: jcollie
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- change ast_settimeout() to honor max rate in edge cases of file playback
(this will make some warning messages go away at the end of playing back
a file)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
account that multiple threads could hold the same rdlock at the same time.
As such, it expected that when a thread released a lock that it must have
been the last to acquire the lock as well. Erroneous error messages would
be sent to the console stating that a thread was attempting to unlock a lock
it did not own.
Now all threads are examined to be sure that the message is only printed
when it is supposed to be printed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ys
Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.
I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c
I did a simple sanity test to make sure the code doesn't
mess things up in general.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert the last part of channel.c over to use the timing API. This would
not have made a difference when using the dahdi timing module. I noticed
it when trying to use another timing source. Oops. :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines
(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia
Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.
The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.
The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.
The T option was added to forkCDR to force
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.
The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via
email, irc, etc, over the past months/year)
The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.
Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!
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This commit pulls in a batch of improvements and additions to the event API.
Changes include:
- the ability to dynamically build a subscription. This is useful if you're
building a subscription based on something you receive from the network,
or from options in a configuration file.
- Add tables of event types and IE types and the corresponding string
representation for implementing text based protocols that use these
events, for showing events on the CLI, reading configuration that
references event information, among other things.
- Add a table that maps IE types and the corresponding payload type.
- an API call to get the total size of an event
- an API call to get all events from the cache that match a subscription
- a new IE payload type, raw, which I used for transporting the Entity ID in
my code for handling distributed device state.
- Code improvements to reduce code duplication
- Include the Entity ID of the server that originated the event in every event
- an additional event type, DEVICE_STATE_CHANGE, to help facilitate distributed
device state. DEVICE_STATE is a state change on one server, DEVICE_STATE_CHANGE
is the aggregate device state change across all servers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit breaks out some logic from pbx.c into a simple API. The hint
processing code had logic for taking the state from multiple devices and
turning that into the state for a single extension. So, I broke this out
and made an API that lets you take multiple device states and determine
the aggregate device state. I needed this for some core device state changes
to support distributed device state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the checks for the CLM and EVT services from the SAForum AIS. I'm going to work
on merging in changes from this branch in pieces.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
hold tracking information for mutexes. Now, the "core show locks" output
will output information about who is holding a rwlock when a thread is
waiting on it.
(closes issue #11279)
Reported by: ys
Patches:
trunk_lock_utils.v8.diff uploaded by ys (license 281)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a deadlock is detected, then the typical lock information will be
printed along with a backtrace of the stack for the offending threads.
Use of this requires compiling with DETECT_DEADLOCKS and having glibc
installed.
Furthermore, issuing the "core show locks" CLI command will print the
normal lock information as well as a backtraces for each lock. This
requires that DEBUG_THREADS is enabled and that glibc is installed.
All the backtrace features may be disabled by running the configure
script with --without-execinfo as an argument
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Until this change, all verbose messages in Asterisk's log files reported logger.c
as the source of the message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sean Bright
Tilghman Lesher
Steve Murphy
Kevin P. Fleming
Mark Michelson
Russell Bryant
Olle Johansson
Terry Wilson
Philippe Sultan
Michiel van Baak
Brett Bryant
Doug Bailey
Jeff Peeler