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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135265 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134922 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@134353 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@134125 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133945 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133770 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133665 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133651 f38db490-d61c-443f-a65b-d21fe96a405b
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. :)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133575 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@133566 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@133448 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133299 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133171 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132964 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@132390 f38db490-d61c-443f-a65b-d21fe96a405b
removed early (before the routine to load the configuration was
finished) because a variable wasn't initialized.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132242 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132206 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@132050 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@131717 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@131445 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: mnicholson
Spent most of the day on this bug, and the
solution was so simple. Just had to find and
understand the problem.
The problem was, that the routine to copy
the existing switches, includes, and ignorepats
from the old context to the new one, wasn't
getting called when the context is already
existent. (In other words, if AEL is adding
a new context to the mix, they get copied,
but if pbx_config already defined a context,
then the copy wasn't happening. This made
no sense, so I moved the call to copy the
includes & etc, no matter the case.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131129 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131044 f38db490-d61c-443f-a65b-d21fe96a405b
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130744 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@130297 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines
Fix a typo in audiohook_read_frame_both.
While this change has not been proven to fix any
specific issue, it is incorrect and could cause
unforeseen problems.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130174 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@130145 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines
fix a flaw found while experimenting with structure alignment and padding; low-fence checking would not work properly on 64-bit platforms, because the compiler was putting 4 bytes of padding between the fence field and the allocation memory block
added a very obvious runtime warning if this condition reoccurs, so the developer who broke it can be chastised into fixing it :-)
........
r129967 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines
simplify calculation
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129968 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines
Fix a problem where inbound rfc2833 audio would be sent to the
core instead of being P2P bridged. When the core regenerated
the rfc2833 packet for the outbound leg, the SSRC would be different
than the RTP audio on the call leg causing DTMF detection issues on
the far end.
(closes issue #12955)
Reported by: tonyredstone
Patches:
dynamic_rtp.patch uploaded by tsearle (license 373)
Tested by: tonyredstone
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129437 f38db490-d61c-443f-a65b-d21fe96a405b
first SRV result from DNS before processing weights and priorities and
dns_parse_answer wouldn't report that there were no records in DNS
unless a failure occured. Also fixed a bug where dnsmgr_refresh would
report that a entry was being changed when ast_gethostbyname had failed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129156 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@128240 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@128197 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128027 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127793 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127401 f38db490-d61c-443f-a65b-d21fe96a405b
callback. This change differs from commit 127113 in that now the
channel is not set to AST_STATE_UP until after the answer callback.
(closes issue #12924)
Reported by: snyfer
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127157 f38db490-d61c-443f-a65b-d21fe96a405b
- 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: http://svn.digium.com/svn/asterisk/trunk@125332 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125138 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@124127 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@123546 f38db490-d61c-443f-a65b-d21fe96a405b
a memory leak when running the astobj2 test CLI command. After
searching, it appears the leak was in the command handler itself.
Each object was allocated (recount = 1) and then linked into
a container (refounct = 2). Then at the end of the function,
the container was unreffed, causing all the objects to have
their refcount decremented by one, leaving the refcount for
all objects allocated in that function at 1. I've now added
an extra unref to the mix so that the refcount equals zero
when the container is unreffed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123526 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines
Fix a memory leak in astobj2 that was pointed out by seanbright. When a container
got destroyed, the underlying bucket list entry for each object that was in the
container at that time did not get free'd.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123272 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@123165 f38db490-d61c-443f-a65b-d21fe96a405b
- 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: http://svn.digium.com/svn/asterisk/trunk@122923 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines
Short circuit the loop in autoservice_run if there are no channels to poll.
If we continued, then the result would be calling poll() with a NULL
pollfd array. While this is fine with POSIX's poll(2) system call, those
who use Asterisk's internal poll mechanism (Darwin systems) would have
a failed assertion occur when poll is called.
(related to issue #10342)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122714 f38db490-d61c-443f-a65b-d21fe96a405b
- 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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122523 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: davidw
Patch by: Corydon76, modified by me to work properly with ParkAndAnnounce app
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122433 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines
Occasionally, the alertpipe loses its nonblocking status, so detect and correct
that situation before it causes a deadlock. (Reported and tested by ctooley
via #asterisk-dev)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122131 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line
Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122128 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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!
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122091 f38db490-d61c-443f-a65b-d21fe96a405b
This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121559 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@121555 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@121501 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@121439 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines
Do not attempt to do emulation if an END digit is received and the length is
less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).
(closes issue #12778)
Reported by: tsearle
Patches:
12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121282 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
........
r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120906 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@120635 f38db490-d61c-443f-a65b-d21fe96a405b
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120602 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines
(closes issue #11594)
Reported by: yem
Tested by: yem
This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120174 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@120171 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@120064 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines
Improve CLI command blacklist checking for the command manager action. Previously,
it did not handle case or whitespace properly. This made it possible for blacklisted
commands to get executed anyway.
(closes issue #12765)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119744 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) | 10 lines
Fix a race condition in channel autoservice. There was still a small window of opportunity
for a DTMF frame, or some other deferred frame type, to come in and get dropped.
(closes issue #12656)
(closes issue #12656)
Reported by: dimas
Patches:
v3-12656.patch uploaded by dimas (license 88)
-- with some modifications by me
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119157 f38db490-d61c-443f-a65b-d21fe96a405b
or you get undefined variable errors when you are compiling under the influence
of MTX_PROFILE
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119074 f38db490-d61c-443f-a65b-d21fe96a405b
before we enter manage_parkinglot.
This will get rid of CLI warnings like:
__ast_read: Exception flag set on 'SIP/<NUMBER>-<ID>', but no exception handler
(closes issue #12748)
Reported by: nreinartz
Patches:
asterisk-multiparking_initialize_filedescr_sets-0.0.1.patch uploaded by nreinartz (license 452)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119015 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | 46 lines
(closes issue #10668)
(closes issue #11721)
(closes issue #12726)
Reported by: arkadia
Tested by: murf
These changes:
1. revert the changes made via bug 10668;
I should have known that such changes,
even tho they made sense at the time,
seemed like an omission, etc, were actually
integral to the CDR system via forkCDR.
It makes sense to me now that forkCDR didn't
natively end any CDR's, but rather depended
on natively closing them all at hangup time
via traversing and closing them all, whether
locked or not. I still don't completely
understand the benefits of setvar and answer
operating on locked cdrs, but I've seen
enough to revert those changes also, and
stop messing up users who depended on that
behavior. bug 12726 found reverting the changes
fixed his changes, and after a long review
and working on forkCDR, I can see why.
2. Apply the suggested enhancements proposed
in 10668, but in a completely compatible
way. ForkCDR will behave exactly as before,
but now has new options that will allow some
actions to be taken that will slightly
modify the outcome and side-effects of
forkCDR. Based on conversations I've had
with various people, these small tweaks
will allow some users to get the behavior
they need. For instance, users executing
forkCDR in an AGI script will find the
answer time set, and DISPOSITION set,
a situation not covered when the routines
were first written.
3. A small problem in the cdr serializer
would output answer and end times even
when they were not set. This is now
fixed.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118880 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) | 6 lines
When showing an error message for a command, don't shorten the command output,
as it tends to confuse the user (it's fine for suggesting other commands,
however).
Reported by: seanbright (on #asterisk-dev)
Fixed by: me
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118554 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) | 8 lines
NULL character should terminate only commands back to the core, not log
messages to the console.
(closes issue #12731)
Reported by: seanbright
Patches:
20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
Tested by: seanbright
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118466 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@118173 f38db490-d61c-443f-a65b-d21fe96a405b
requires manager authentication for any POST's being sent to the server as well to help secure uploads.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118161 f38db490-d61c-443f-a65b-d21fe96a405b
- 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: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
display this information in the "core show settings" CLI command. This is
useful if you want to verify that you're running a build with DONT_OPTIMIZE,
DEBUG_THREADS, etc.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117756 f38db490-d61c-443f-a65b-d21fe96a405b
Until this change, all verbose messages in Asterisk's log files reported logger.c
as the source of the message.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117693 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) | 7 lines
1) Don't print the verbose marker in front of every message from ast_verbose()
being sent to remote consoles.
2) Fix pbx_gtkconsole to filter out the verbose marker.
(related to issue #12702)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117508 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) | 6 lines
Don't display the verbose marker for calls to ast_verbose() that do not include
a VERBOSE_PREFIX in front of the message.
(closes issue #12702)
Reported by: johnlange
Patched by me
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117481 f38db490-d61c-443f-a65b-d21fe96a405b
a configuration file. As it was, only the first did so. This led to
a problem if the included file was changed (but not the configuration
file which includes it) and the second source file attempted to reload
the configuration. It would not see that the included file had changed.
In this particular example, res_phoneprov and chan_sip both loaded
sip.conf, which included a file call sip.peers.conf. Since res_phoneprov
was the first to load sip.conf, only it cached the fact that sip.conf
included sip.peers.conf. If sip.peers.conf were changed and sip.conf were
not and a sip reload were issued (meaning that chan_sip attempts to
reload sip.conf only if it and its included files have changed) the changes
made to sip.peers.conf would not be seen and therefore no action would be
taken.
(closes issue #12693)
Reported by: marsosa
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117367 f38db490-d61c-443f-a65b-d21fe96a405b
last match, and possibly skip empty fields. The function is useful
(and used here) when a form submits multiple 'Action' fields to the
Manager.
This change slightly modifies the current behaviour, but only in the
case the user supplies multiple 'Action: ' lines and the first
ones are empty, so the change is totally harmless.
+ Fix style on a couple of "if (displayconnects)" statements;
+ Expand a bit the 'Manager Test' interface, to make it slightly
more user friendly. But also comment that the HTML should not
be embedded in the C source.
None of this stuff needs to be applied to 1.4.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117297 f38db490-d61c-443f-a65b-d21fe96a405b
same name in http queries, which might confuse the manager.
Replace calls to ast_uri_decode() with a local function that also
replaces '+' with ' ', as this is the normal encoding for
spaces in http requests.
This allows passing cli commands to the manager through the
http interface.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117295 f38db490-d61c-443f-a65b-d21fe96a405b
of platform/compiler-dependent warnings when handing
struct timeval fields, both reading and printing them.
It is a lost battle to handle the different ways struct timeval
is handled on the various platforms and compilers, so try
to be pragmatic and go through int/long which are universally
supported.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116557 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines
A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS
is enabled in menuselect, the actual origin of channel locks is obscured
by the fact that all channel locks appear to happen in the function
ast_channel_lock(). This code change redefines ast_channel_lock to be a
macro which maps to __ast_channel_lock(), which then relays the proper
file name, line number, and function name information to the core lock
functions so that this information will be displayed in the case that
there is some sort of locking error or core show locks is issued.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116089 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) | 5 lines
Fix an issue that I noticed in autoservice while mmichelson and I were debugging
a different problem. I noticed that it was theoretically possible for two threads
to attempt to start the autoservice thread at the same time. This change makes the
process of starting the autoservice thread, thread-safe.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116001 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May 2008) | 7 lines
If a thread holds no locks, do not print any information on the thread when issuing
a core show locks command. This will help to de-clutter output somewhat.
Russell said it would be fine to place this improvement in the 1.4 branch, so that's
why it's going here too.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115737 f38db490-d61c-443f-a65b-d21fe96a405b
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115588 f38db490-d61c-443f-a65b-d21fe96a405b
new paramater. The new options for ENUM* functions include 'u', 's', 'i', and 'd' which return the full uri, trigger isn specific rewriting, look
for branches into an infrastructure enum tree, or do a direct dns lookup of a number respectively. The new paramater for TXCIDNAME adds a
zone-suffix argument for looking up caller id's in DNS that aren't e164.arpa.
This patch is based on the original code from otmar, modified by snuffy, and tested by jtodd, me, and others.
(closes issue #8089)
Reported by: otmar
Patches:
20080508_bug8089-1.diff
- original code by otmar (license 480),
- revised by snuffy (license 35)
Tested by: oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115584 f38db490-d61c-443f-a65b-d21fe96a405b
to use doubly linked lists. The schedule() function had an optimization that
had it try to guess which direction would be better for the traversal to insert
the task into the scheduler queue. However, if the code chose the path where
it traversed the queue in reverse, and the result was that the task should be
at the head of the queue, then the code would actually put it at the tail,
instead.
(Problem found by bbryant, debugged and fixed by bbryant and me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115537 f38db490-d61c-443f-a65b-d21fe96a405b
being #included twice. This was due to the fact that #exec created a temporary file which
was then #included. The name of the temporary file was the name of the #exec'd file, with
the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements
of the same file were reached in the same second, then the result was that the temporary files
would have duplicate names. To resolve this, the temporary file now has microsecond resolution
for the timestamp portion.
(closes issue #12574)
Reported by: jmls
Patches:
12574.patch uploaded by putnopvut (license 60)
Tested by: jmls, putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115329 f38db490-d61c-443f-a65b-d21fe96a405b
