https://origsvn.digium.com/svn/asterisk/trunk
........
r123165 | murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines
(closes issue #12689)
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/branches/1.6.0@123173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines
Move compatibility options into asterisk.conf, default them to on for upgrades,
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/branches/1.6.0@120172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r114553 | murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines
(closes issue #12469)
Reported by: triccyx
I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff)
but finally, I got it. The problem appears to be that the extension is being added in
replace mode, which kinda assumes that the pattern trie has been formed, when in fact,
in this case, it was not. The checks being done are not nec. when the tree is not yet
formed, as changes like this will be summarized when the trie is formed in the future.
I tested the fix, and the crash no longer happens. Feel free to open the bug again if
this fix doesn't cure the problem.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r114146 | murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines
These changes:
a. fix a self-found problem with SPAWN-ing an extension,
where matches were not being found
b. correct some wording in a comment
c. Add some debug for future debugging.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r113836 | mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14 lines
There was a subtle logical difference between 1.4 and trunk with regards to how timeouts
were handled. In 1.4, if the absolute timeout were reached on a call, no matter what
the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T'
extension or hangup otherwise. The rearrangement of this function in trunk made this check
only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned
1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite
loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will
behave as they did in 1.4
(closes issue #11550)
Reported by: pj
Tested by: putnopvut
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@113837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r112289 | murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines
(closes issue #12298)
Reported by: falves11
Patches:
12298.patch1 uploaded by murf (license 17)
Tested by: murf
I have hopes that the changes made over the last few days will
finalize and solidify this code. While there are bound to be
small tweaks still needed, I feel that the job (at last) is
somewhat completed. Finally, I had a chance to comprehend how
the scoring of extension patterns was done in the previous
version, and I've come very close to using the exact same
criteria in the new pattern matching code. The left-right
sorting is now replicated in the trie structure itself, such
that the first match found will the 'best' match. Compared
the results against 1.4 for several extensions. Replicated
falves11's setup and it works. Used some devious patterns
provided by jsmith, supplemented with a few of my own.
Looks good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@112299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, 27 Mar 2008) | 17 lines
Merged revisions 111391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines
These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@111411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008) | 16 lines
Merged revisions 107161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines
Fix another bug specifically related to asynchronous call origination. Once the
PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread. We can no longer access it in this
code. If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.
(inspired by BE-386)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008) | 17 lines
Merged revisions 107158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines
Fix some bugs related to originating calls. If the code failed to start a PBX
on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed. This caused weird memory corruption and
crashes to occur.
(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar 2008) | 16 lines
Merged revisions 106437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines
Quell an annoying message that is likely to print every single time that
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.
Thanks to joetester on IRC for pointing this out
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@106442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb 2008) | 9 lines
Make pbx_exec pass an empty string into applications, if we get NULL.
This protects against possible segfaults in applications that may try
to use data before checking length (ast_strdupa'ing it, for example)
(closes issue #12100)
Reported by: foxfire
Patches:
12100-nullappargs.diff uploaded by qwell (license 4)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines
Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then
calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is
dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension
into ast_merge_contexts_and_delete (sans the extra lock).
(this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
equivalent to the check done by ast_verb, I wrote a macro, VERBOSITY_LEVEL, which does this
check. I did a quick look in the source and used this macro in some places where option_verbose
was used.
I also converted some verbose messages in logger.c to use ast_verb instead of ast_verbose.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | 7 lines
Make some changes to some additions I made recently for doing channel autoservice
when looking up extensions. This code was added to handle the case where a
dialplan switch was in use that could block for a long time. However, the way
that I added it, it did this for all extension lookups. However, lookups in the
in-memory tree of extensions should _not_ take long enough to matter. So, move
the autoservice stuff to be only around executing a switch.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91011 65c4cc65-6c06-0410-ace0-fbb531ad65f3