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(closes issue #14049)
Reported by: kshumard
Patches:
doc.tex.qos.tex.patch uploaded by kshumard (license 92)
(Slight modifications by seanbright)
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use, so tell the core that we don't know the devstate and have it ask us for it.
(closes issue #13525)
Reported by: pj
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) | 3 lines
Oops, inverted logic for a strcasecmp check. Pointed out by mmichelson, thanks!
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single memory allocation we can't just use the existing CLI alias structure. We have to destroy all existing ones and then create new ones.
(closes issue #14054)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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channel variables, and in any case, was redundant;
pbx_substitute_variables_helper ALREADY does substitution for global
variables.
(closes issue #13327)
Reported by: pj
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r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines
(closes issue #13229)
Reported by: clegall_proformatique
Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams.
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r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines
Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
(closes issue #12560)
Reported by: vsauer
Patches:
patch001.diff uploaded by ramonpeek (license 266)
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r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines
When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599)
Reported by: hjourdain
Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
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(closes issue #13969)
Reported by: jtodd
Patches:
20081205__bug13969.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak, eliel
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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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r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines
Add missing documentation to misdn.txt
(closes issue #14052)
Reported by: festr
Patches:
misdn.txt.patch uploaded by festr (license 443)
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r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines
Revert fix for issue 13570. It has caused more problems than
it helped to fix.
(closes issue #13783)
Reported by: navkumar
(closes issue #14025)
Reported by: ffs
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(closes issue #14051)
Reported by: ys
Patches:
res_http_post.c.diff uploaded by ys (license 281)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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transport being used, not the port for the remote server.
(closes issue #13633)
Reported by: performer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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OpenBSD uses an old version of gcc which throws an error
if you use a macro that's not #defined
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(closes issue #14032)
Reported by: bkruse
Patches:
14032.patch uploaded by bkruse (license 132)
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r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines
Oops, should be "tz", not "zonetag".
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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.
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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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r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines
We appear to have documented tz= in the [general] section of voicemail.conf,
without actually having implemented it. Oops.
(Reported by Olivier on the -users list)
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r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines
Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing.
(closes issue #14005)
Reported by: ddl
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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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r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines
Fix double declaration of 'x' on the PPC platform.
(closes issue #14038)
Reported by: ffloimair
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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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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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r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines
Take video into account when early bridging RTP.
(closes issue #13535)
Reported by: davidw
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r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 line
Previous fix used ast_malloc and ast_copy_string and messed up the standalone stuff. Fixed.
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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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r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines
Allow DISA to handle extensions that start with #.
(closes issue #13330)
Reported by: jcovert
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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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was not being freed until the
main thread ends. (That is never).
(closes issue #14040)
Reported by: eliel
Patches:
func_odbc.c.patch uploaded by eliel (license 64)
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for ulaw.
(closes issue #14001)
Reported by: henrikw
Patches:
alw.diff uploaded by henrikw (license 627)
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(closes issue #14027)
Reported by: junky
Patches:
M14027.diff uploaded by junky (license 177)
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milliseconds. collect_digits now gets milliseconds fed to it, not seconds.
(closes issue #14012)
Reported by: dveiga
Patches:
14012.patch uploaded by bkruse (license 132)
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r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines
Make the usereqphone option work again.
(closes issue #13474)
Reported by: mmaguire
Patches:
20080912_bug13474.diff uploaded by mmaguire (license 571)
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rtp stats.
(closes issue #13956)
Reported by: chris-mac
Tested by: chris-mac
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SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
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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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applications:
- VoiceMail()
- VoiceMailMain()
- MailboxExists()
- VMAuthenticate()
functions:
- MAILBOX_EXISTS()
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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.
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