Age | Commit message (Collapse) | Author |
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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
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(closes issue #13082)
Reported by: eliel
Patches:
app_rpt.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008) | 6 lines
Fix a bug in iax2 registration that allowed peers to register with
case-insensitive names (user_cmp_cb and peer_cmp_cb are now both
case-sensitive).
(closes issue #13091)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue #13057)
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r131480 | tilghman | 2008-07-16 16:46:06 -0500 (Wed, 16 Jul 2008) | 2 lines
Apparently, in certain cases, a callno is already destroyed when iax2_destroy is called.
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calculate the number of bytes from a sysinfo structure.
unsigned long.
(closes issue #13057)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008) | 7 lines
Always ensure that the channel's tech_pvt reference is NULL after calling the
destroy callback.
(closes issue #13060)
Reported by: jpgrayson
Patches:
chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license 492)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
(closes issue #13084)
Reported by: elbriga
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we know that it's NULL.
(closes issue #13092)
Reported by: gknispel_proformatique
Patches:
trunk_sqlite_check_vars_null.patch uploaded by gknispel (license 261)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines
Apparently, "thread safety" is important, whatever
that means. :P
(Thanks Russell!)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
(closes issue #13047)
Reported by: festr
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r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines
(closes issue #13090)
Reported by: murf
The problem was that, esoteric as it is, because the hangerupper
context immediately preceded the std-priv-extent macro, that
the checking code accidentally would fall from traversing hangerupper
into the std-priv-exten macro, where it would hit the hangerupper
in the 'includes', and proceed into an infinite recursion.
A small fix to traverse into the statements of the context instead
of the context solves this issue.
I also added some commented out printfs for debug, which were pretty
handy in the face of a dorky gdb.
This was a problem around since the package was first written;
but evidently pretty rare in turning up in the field.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(from rev 131206 in the 1.6.0 branch)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Closes issue #13086)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@131129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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released, causing a deadlock. (Reported by mvanbaak in #asterisk-dev,
discovered by bbryant's change to the lock tracking code to yell at you
if a thread exits with a lock still held)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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a member of any queue.
(closes issue #13073)
Reported by: eliel
Patches:
app_queue.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) | 7 lines
remove 4 lines of redundant code.
(closes issue #13080)
Reported by: gknispel_proformatique
Patches:
trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines
Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
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opened when running startup commands from cli.conf
(closes issue #13066)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) | 10 lines
notify the user that dnsmgr refresh wont work when dnsmgr is not enabled.
Previously this command would automagically appear and disappear.
This was confusing.
(closes issue #12796)
Reported by: chappell
Patches:
dnsmgr_refresh_3.diff uploaded by chappell (license 8)
Tested by: russell, chappell, mvanbaak
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it with separate calls
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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declaration (suppresses a warning).
(closes issue #13070)
Reported by: gknispel_proformatique
Patches:
asterisk_trunk_const_static.patch uploaded by gknispel (license 261)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines
Bump up the debug level for a message.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed
a locally modified asterisk.tex, with changes done by this script.
(closes issue #13062)
Reported by: mvanbaak
Patches:
sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak
Feedback from Corydon. Thanks for taking the time to go through this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12956)
Reported by: DEA
Patches:
chan_skinny-linkedlists-v2.txt uploaded by DEA (license 3)
Tested by: DEA, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) | 8 lines
fix memory leak when originate from manager cannot create a thread
(closes issue #13069)
Reported by: gknispel_proformatique
Patches:
asterisk_trunk_action_originate.patch uploaded by gknispel (license 261)
Tested by: gknispel_proformatique, mvanbaak
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008) | 4 lines
Reverting 2 changesets, as it breaks incoming IAX2 calls
(Related to issue #12963)
Reported by: mvanbaak
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This got lost in commit 97634
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r130373 | mvanbaak | 2008-07-12 12:25:52 +0200 (Sat, 12 Jul 2008) | 6 lines
in 1.4 the functions still have | as argument seperator.
This commit fixes the use of RAND in the ael random function.
(closes issue #13061)
Reported by: danpwi
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines
Remove redundant logic
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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.
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r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines
Ensure that a destination callno of 0 will not match for frames that do not
start a dialog (new, lagrq, and ping).
(closes issue #12963)
Reported by: russellb
Patches:
chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
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Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines
Pass the devicestate from an underlying channel up through the Agent channel.
This should make the Agent always report the correct device state, even when
the underlying channel is used for other purposes.
(closes issue #12773)
Reported by: davidw
Patches:
20080710__bug12773.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw
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