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r162671 | murf | 2008-12-10 09:45:01 -0700 (Wed, 10 Dec 2008) | 22 lines
(closes issue #14022)
Reported by: wetwired
Tested by: murf
I checked, and I added a mod to the trunk version
of Asterisk that would make it 8-bit transparent
on 27 Nov 2007, but I made no such updates to
1.4. My best guess is that 1.4 was released, and
it was not appropriate to commit an enhancement.
But I'm going to add the same fixes to 1.4 now,
for the following reasons:
1. wetwired is correct; 1.4 is **mostly** 8-bit
transparent now. This is because the lexical
token forming rules use . in most 'word'
state continuances. It's just the beginning
of a 'word' that is picky.
2. Accepting 8-bit chars in some places and
not others leads to bug reports like this.
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r162670 | mmichelson | 2008-12-10 10:44:37 -0600 (Wed, 10 Dec 2008) | 14 lines
Update to stringfield handling so that side-effects on
parameters are not evaluated multiple times.
An example where this caused a problem was in chan_sip.c, with
the line
ast_string_field_set(p, fromdomain, ++fromdomain);
This patch was originally uploaded to issue #13783 by
jamessan. While the issue was closed for other reasons, this
patch is valid and fixes a separate problem, and is thus
being committed.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines
Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
(closes issue #12983)
Reported by: vt
Patches:
dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
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After the nightly update of the documentation on asterisk.org, I'll post
an update to asterisk-dev with a pointer to the changes. This covers some
release branch and commit policy information. None of this should be a
surprise, since it's just documenting what we have already been doing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines
Remove the test_for_thread_safety() function completely.
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
(closes issue #12471)
Reported by: mthomasslo
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line
In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines
If we fail to start a thread for the pbx to run in, we need to
be sure to decrease the number of active calls on the system.
This fix may relate to ABE-1713, but it is not certain yet.
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r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
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r162071 | tilghman | 2008-12-09 11:07:50 -0600 (Tue, 09 Dec 2008) | 3 lines
For some reason, after a distclean, gcc started returning
'value computed is not used'. Fixing (for --enable-dev-mode).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines
(closes issue #14019)
Reported by: ckjohnsonme
Patches:
14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf
This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.
32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.
Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.
I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.
And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.
Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.
Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!
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r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines
Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old channel to the new
one. However, it did nothing with the group settings that were already on the
new channel. This patch removes all group settings that already existed on the
new channel.
I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category. However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.
(closes AST-152)
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- Avoid printing the name of an argument in the [Arguments] tag if there is no description
for that argument.
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add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.
This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed
by Mark M. on IRC.
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when a save occurs to a different filename, everything goes to the alternate
filename, instead of appending to the original. This is important for the
AMI command UpdateConfig.
(closes issue #13301)
Reported by: trevo
Patches:
20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
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