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r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines
Merged revisions 280341 via svnmerge from
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r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines
Fix a dsp structure leak occuring when a local channel is put into a meetme
conference, then masquaraded away.
ABE-2422
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related to r280302
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r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines
Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported.
ABE-2229
Review: https://reviewboard.asterisk.org/r/813/
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Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699.
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r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 Jul 2010) | 2 lines
Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options.
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r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) | 6 lines
Work around some silly behavior on BSD.
A non-zero exit from a subshell should make the build fail.
(closes issue #17621)
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r280227 | qwell | 2010-07-28 14:54:54 -0500 (Wed, 28 Jul 2010) | 1 line
Add sha1sum-sh in case there is no util on the system.
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r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) | 9 lines
Remove unnecessary subshells. Attempt to make checksumming work.
Also improves readability.
(issue #17621)
Reported by: bjm
Review: https://reviewboard.asterisk.org/r/808/
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r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines
Plug a reference leak in app_queue when adding members dynamically.
(closes issue #17738)
Reported by: bobwienholt
Patches:
issue17738.patch uploaded by bobwienholt (license 950)
Tested by: bobwienholt, seanbright
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r280089 | lmadsen | 2010-07-28 08:51:16 -0500 (Wed, 28 Jul 2010) | 9 lines
Merged revisions 280088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line
Update help text to be less confusing.
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(closes issue #17732)
Reported by: pabelanger
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The version of libedit that is bundled with asterisk is old and has some bugs.
This patch updates the bundled version of libedit within asterisk, and also
updates asterisk to use the system libedit instead if one is available (and
pkg-config is available). This review integrates several patches from other
users specifically kkm and tzafrir.
(closes issue #15929)
Reported by: kkm
Patches:
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
(issue #16858)
Reported by: jw-asterisk
(closes issue #17039)
Reported by: tzafrir
Patches:
0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/807/
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This option enables the proper compiler flags for tracking code coverage, which
is useful along side automated testing.
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r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
Merged revisions 279945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
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This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
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The code was written in a way that did a bad job of
parsing the port out of a URI. Specifically, it would
do badly when dealing with an IPv6 address. In this
particular scenario, there was no value from parsing
the port out, so I just removed that logic. And while
I was messing around in the function, I changed some
variable names to be more descriptive.
(closes issue #17661)
Reported by: oej
Patches:
17661.diff uploaded by mmichelson (license 60)
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r279883 | qwell | 2010-07-27 12:54:54 -0500 (Tue, 27 Jul 2010) | 1 line
Add SHA1SUM to configure, since we require it for sounds/
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r279852 | qwell | 2010-07-27 11:48:41 -0500 (Tue, 27 Jul 2010) | 1 line
Remove aptly-named EMPTY and BS vars, since they aren't used anymore.
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r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) | 1 line
Simply sounds/Makefile some more.
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when using getaddrinfo
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Review: https://reviewboard.asterisk.org/r/804
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r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines
Fix bad behavior of dynamic_exclude_static option in sip.conf.
We were attempting to create a contactdeny rule based on the peer's
IP address before the peer's IP address had been set. By moving the
processing further down in the function, we can ensure stuff works
as we expect for it to.
(closes issue #17717)
Reported by: mmichelson
Patches:
17717.patch uploaded by mmichelson (license 60)
Tested by: DennisD
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(closes issue #17693)
Reported by: iasgoscouk
Patches:
issue17693.patch uploaded by pabelanger (license 224)
Tested by: iasgoscouk
Review: https://reviewboard.asterisk.org/r/803/
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Otherwise, ast_tcptls_server_start() will still start http.
(closes issue #17708)
Reported by: pabelanger
Patches:
http.patch uploaded by pabelanger (license 224)
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r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines
Really fix sounds Makefile (and make it readableish).
There was a rather large syntax error that should have caused ALL versions of GNU make to fail.
I don't know how it worked.
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r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26 Jul 2010) | 2 lines
Dunno why this worked on my machine, but it works better this way.
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r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | 13 lines
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573
(closes issue #13573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
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r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) | 13 lines
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573 [^]
(closes issue 0013573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
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This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
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(closes issue #17715)
Reported by: vrban
Patches:
17715.patch uploaded by pabelanger (license 224)
Tested by: vrban
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r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) | 2 lines
Use a special Makefile for noobs who still have GNU Make 3.80.
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A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
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r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, 26 Jul 2010) | 5 lines
Expand the correct value within AST_OPTION_ONLY.
(closes issue #17703)
Reported by: stuarth
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ast_format_str_reduce() at load time.
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(closes issue #17304)
Reported by: jnemeth
Patches:
20100507__issue17304.diff.txt uploaded by tilghman (license 14)
Tested by: jnemeth
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(closes issue #17704)
Reported by: vrban
Patches:
issue17704.patch uploaded by pabelanger (license 224)
Tested by: vrban
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(closes issue #17695)
Reported by: pabelanger
Patches:
Makefile.diff uploaded by pabelanger (license 224)
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(closes issue #17677)
Reported by: outcast
Patches:
issue0017677.patch uploaded by pabelanger (license 224)
Tested by: elguero
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Otherwise, 'manager show settings' will generate errors
if manager is not enabled.
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r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines
Merged revisions 279206 via svnmerge from
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r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines
SIP promiscuous redirect could fail to dial the redirect.
The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
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