Age | Commit message (Collapse) | Author |
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(thanks for the report, Doug), and 2) weren't actually being appended to the list of profile variables because build_profile was called before the list was populated. Also needed to free the contents returned by load_file().
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr 2008) | 11 lines
Fix a race condition that may happen between a sip hangup
and a "core show channel" command. This patch adds locking
to prevent the resulting crash.
(closes issue #12155)
Reported by: tsearle
Patches:
show_channels_crash2.patch uploaded by tsearle (license 373)
Tested by: tsearle
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(closes issue #12422)
Reported by: nito
Patches:
res_config_ldap_result_check_patch.diff uploaded by nito (license 340)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines
Fix 1.4 build when LOW_MEMORY is enabled.
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channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines
Be sure that we're not about to set bridgepvt NULL prior to dereferencing it.
(closes issue #11775)
Reported by: fujin
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(closes issue #12401)
Reported by: davevg
Patches:
astcli.diff2 uploaded by davevg (license 209)
Tested by: davevg, Corydon76
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines
Only try to prefix language if we are not using an absolute path (suffix it otherwise).
en/var/lib/asterisk/sounds/blah.gsm is a very silly path.
(closes issue #12379)
Reported by: kuj
Patches:
12379-absolutepath.diff uploaded by qwell (license 4)
Tested by: kuj, qwell
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r114032 | file | 2008-04-10 12:58:39 -0300 (Thu, 10 Apr 2008) | 6 lines
Forgot the 1.4 branch for russian language fix.
(closes issue #12404)
Reported by: IgorG
Patches:
voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
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r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines
Create the directory where name recordings will go if it does not exist.
(closes issue #12311)
Reported by: rkeene
Patches:
12311-mkdir.diff uploaded by qwell (license 4)
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(closes issue #12404)
Reported by: IgorG
Patches:
voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12409)
Reported by: candlerb
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines
Don't add custom URI options if they don't exist OR they are empty.
(closes issue #12407)
Reported by: homesick
Patches:
uri_options-1.4.diff uploaded by homesick (license 91)
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(closes issue #12396)
Reported by: tcalosi
Patches:
12396.patch uploaded by putnopvut (license 60)
Tested by: tcalosi
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines
We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.
(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines
If the [csv] section does not exist in cdr.conf, then an unload/load sequence
is needed to correct the problem. Track whether the load succeeded with a
variable, so we can fix this with a simple reload event, instead.
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(closes issue #11901)
Reported by: pj
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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
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(closes issue #12224)
Reported by: DEA
Patches:
chan_skinny-debug-log.txt uploaded by DEA (license 3)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines
If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario.
(closes issue #12385)
Reported by: viraptor
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(closes issue #12254)
Reported by: andrew
Patches:
bug-12253.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines
If Asterisk receives a 488 on an INVITE (not a reinvite), then
we should not send a BYE.
(closes issue #12392)
Reported by: fnordian
Patches:
chan_sip.patch uploaded by fnordian (license 110) with small modification from me
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(closes issue #11954)
Reported by: johan
Patches:
app_dial_rev104031.patch uploaded by johan (license 334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12390)
Reported by: tzafrir
Patches:
astcli_fixes.diff uploaded by tzafrir (license 46)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines
Initialize fr->cacheable to make valgrind happy
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asterisk -rx)
(closes issue #12389)
Reported by: davevg
Patches:
astcli uploaded by davevg (license 209)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r113507 | mmichelson | 2008-04-08 14:07:38 -0500 (Tue, 08 Apr 2008) | 8 lines
Fix potential buffer overflow that could happen if more than 100 announce files
were specified when calling ParkAndAnnounce. This overflow is not exploitable remotely
and so there is no need for a security advisory.
(closes issue #12386)
Reported by: davidw
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r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line
Add a little more that is required for previously added devices.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines
Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975.
Thanks to Greg Oliver for providing me the required information.
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be used by modules.
(closes issue #12384)
Reported by: fnordian
Patches:
features.patch uploaded by fnordian (license 110)
(patch modified by me, to give FEATURE_RETURN_* an AST_ prefix)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line
Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing.
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r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) | 6 lines
Add security note on astgenkey's manpage.
(closes issue #12373)
Reported by: lmamane
Patches:
20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
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r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines
Move check for still-bridged channels out a little further, to avoid possible
deadlocks. (Closes issue #12252)
Reported by: callguy
Patches:
20080319__bug12252.diff.txt uploaded by Corydon76 (license 14)
Tested by: callguy
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r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines
If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal
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r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines
Merged revisions 113012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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while.
The variable name "flag" to distinguish between whether a message is being forwarded or
is new is not a helpful name. The newly added doxygen documentation to app_voicemail is
tremendously helpful, but I still just...hate this variable name. I think is_new_message
is more indicative of what its purpose is.
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r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines
Force ast_mktime() to check for DST, since strptime(3) does not.
(Closes issue #12374)
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r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines
Allow playback with noanswer (and add earlyrtp option).
(closes issue #9077)
Reported by: pj
Patches:
earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn
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r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines
This fix prevents a deadlock that was experienced in chan_local. There was
deadlock prevention in place in chan_local, but it would not work in a specific
case because the channel was recursively locked. By unlocking the channel prior
to calling the generator's generate callback in ast_read_generator_actions(), we
prevent the recursive locking, and therefore the deadlock.
(closes issue #12307)
Reported by: callguy
Patches:
12307.patch uploaded by putnopvut (license 60)
Tested by: callguy
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r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines
(closes issue #12362)
(closes issue #12372)
Reported by: vinsik
Tested by: tecnoxarxa
This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.
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(closes issue #12370)
Reported by: srt
Patches:
asterisk-12370.diff uploaded by srt (license 378)
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quick crash of asterisk, and a road to a DOS attack if left unfixed.
Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes.
I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short.
Instead of freeing the struct and nulling the pointer, it now just resets it, because this
ast_str is expected by the calling routine to still be there after handle_request_do() returns.
This appears to fix the crash. I assume that it was introduced with ast_str's being adopted. It's a subtle and easy-to-miss sort of problem.
I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well;
no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do...
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