https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 Nov 2007) | 8 lines
Remove some checks to see if locks are initialized from the non-DEBUG_THREADS
versions of the lock routines. These are incorrect for a number of reasons:
- It breaks the build on mac.
- If there is a problem with locks not getting initialized, then the proper
fix is to find that place and fix the code so that it does get initialized.
- If additional debug code is needed to help find the problem areas, then this
type of things should _only_ be put in the DEBUG_THREADS wrappers.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines
When traversing the list of channel variables here in transmit_invite(), the
asterisk channel must be locked, as this data may change at any time.
(I have seen numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists. So, I am finding and
fixing some places where channel variables are handled improperly.)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
namely main/Makefile .
I am unclear where decisions on the build environment (CFLAGS,
LDFLAGS, LIBS and so on) should be made - right now they are
split here and there.
As a first step in cleaning up this situation, i am trying to at
least collect all instances of each variable in one place.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | 7 lines
Merge changes from asterisk/team/kpfleming/SRV-priority-handling
Previously, the SRV record support in Asterisk was broken. There was no
guarantee on what record Asterisk would choose to actually use. This set of
changes improves the situation by ensuring that Asterisk will choose the
highest priority record.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | 20 lines
Merge the last bit of changes from asterisk/team/russell/readq-1.4
The issue here is that the channel frame readq handling got broken when the
code was converted to use the linked list macros. It caused corruption of the
list head and tail pointers. So, I fixed up the usage of the linked list
macros and in passing, simplified the code. I also documented what the code
is doing, as it was a bit difficult to figure out at first.
This bug showed itself with crashes showing messed up head/tail pointers for
the readq. However, there are a couple of crashes that aren't quite as obvious,
but I think may be related. So, if your bug gets closed by this commit, but
you still have a problem, please reopen or create a new bug report.
(closes issue #10936)
(closes issue #10595)
(closes issue #10368)
(closes issue #11084)
(closes issue #10040)
(closes issue #10840)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I am preserving the behaviour on *.adsi files,
i.e. overwrite anything there without making a backup.
However I am not sure that this is the intended behaviour.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | 5 lines
Fix up datastore handling in ast_do_masquerade(). The code is intended to move
any channel datastores from the old channel to the new one. However, it did
not use the linked list macros properly to accomplish the task. The existing
code would only work if there was only a single datastore on the old channel.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
too many times.
On passing, add some comments and fix indentation a bit.
On passing, i suspect that the following pattern is wrong
%.eoo: %.o
but in case it will be fixed in a later commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines
If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive.
(closes issue #5747)
Reported by: chandi
Patches:
infofix-81430-1.patch uploaded by IgorG (license 20)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11147)
........
r88283 | qwell | 2007-11-02 11:51:08 -0500 (Fri, 02 Nov 2007) | 4 lines
We need to make sure to specify a language to ast_fileexists, otherwise it may fail for anything besides en
Issue 11147, fix discovered by both citats and myself (independently), with input from Corydon76
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
