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git.ipfire.org Git - thirdparty/asterisk.git/log
Russell Bryant [Tue, 10 Aug 2010 17:47:13 +0000 (17:47 +0000)]
Merged revisions 281566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines
Reset visible indication after answer.
(closes issue #17641)
Reported by: klaus3000
Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65)
Tested by: schmidts
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David Vossel [Mon, 9 Aug 2010 20:46:50 +0000 (20:46 +0000)]
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the
peers_by_ip container. This made it impossible to be removed
from the container as the addr is the key used by the container
to find the peer.
(closes issue #17774)
Reported by: kkm
Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281430
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Jeff Peeler [Mon, 9 Aug 2010 20:07:29 +0000 (20:07 +0000)]
Merged revisions 281390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines
Prevent loss of Caller ID information set on local channel after masquerade.
Caller ID set on the channel before a masquerade occurs when using a local
channel would cause the information to be lost. The problem was that the
information was set on a channel destined to be hung up. The somewhat confusing
fix is to detect if any Caller ID has been set on the channel and if so
preswap the Caller ID data so that basically the masquerade puts the data back.
(closes issue #17138)
Reported by: kobaz
Review: https://reviewboard.asterisk.org/r/847/
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David Vossel [Fri, 6 Aug 2010 21:37:29 +0000 (21:37 +0000)]
Blocked revisions 281185 via svnmerge
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r281185 | dvossel | 2010-08-06 16:34:38 -0500 (Fri, 06 Aug 2010) | 20 lines
chan_sip: fixes provisional keepalive scheduled item crash
There is a scheduler item in chan_sip that keeps sending the
last provisional message in response to an INVITE Request for
a period of time until a final response to that INVITE is
sent. Because of the way this scheduler item works, it requires
a reference to a sip_pvt pointer to work properly. The problem
with this is that it is currently possible (but rare) for the
sip_pvt to get destroyed and that scheduler item to still
exist. When this occurs, the scheduler event fires and attempts
to access a freed sip_pvt which causes a crash.
(closes issue #17497)
Reported by: anonymouz666
Patches:
keepalive_diff_1.4_v2.diff uploaded by dvossel (license 671)
Review: https://reviewboard.asterisk.org/r/849/
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Russell Bryant [Thu, 5 Aug 2010 13:11:32 +0000 (13:11 +0000)]
Cleanup default option value handling for cdr.conf [general].
The default values would differ depending on whether or not cdr.conf exists.
That is no longer the case.
Apply a default value to the unanswered option.
Define all default values as named constants.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281051
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Tilghman Lesher [Thu, 5 Aug 2010 07:40:47 +0000 (07:40 +0000)]
Merged revisions 280982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines
Change context lock back to a mutex, because functionality depends upon the lock being recursive.
(closes issue #17643)
Reported by: zerohalo
Patches:
20100726__issue17643.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo
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Tilghman Lesher [Tue, 3 Aug 2010 20:52:20 +0000 (20:52 +0000)]
Merged revisions 280811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines
Prevent DAHDI channels from overriding the callerid, once it's been set by the user.
(closes issue #16661)
Reported by: jstapleton
Patches:
20100414__issue16661.diff.txt uploaded by tilghman (license 14)
20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: jstapleton
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Tilghman Lesher [Tue, 3 Aug 2010 18:39:28 +0000 (18:39 +0000)]
Document -B and -W flags and regenerate manpage from sgml
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280739
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Tilghman Lesher [Mon, 2 Aug 2010 21:26:11 +0000 (21:26 +0000)]
Allow the pipe, but also allow the comma
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280671
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Jeff Peeler [Mon, 2 Aug 2010 21:14:20 +0000 (21:14 +0000)]
Change SIP NOTIFY requests to expect a response so authentication will work.
This changes the request to be sent with the transmit type XMIT_RELIABLE so that
sip_ack doesn't return false and cause the 401 to be ignored in cases where
authentication is required.
(closes issue #14255)
Reported by: zktech
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Tilghman Lesher [Thu, 29 Jul 2010 21:07:03 +0000 (21:07 +0000)]
Off-by-one error
(closes issue #17590)
Reported by: atis
Patches:
20100729__issue17590.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280556
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David Vossel [Thu, 29 Jul 2010 20:42:29 +0000 (20:42 +0000)]
fixes wrong SRV query for TLS connection
(closes issue #17612)
Reported by: marcelloceschia
Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079)
chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079)
Tested by: marcelloceschia, st, pabelanger
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David Vossel [Thu, 29 Jul 2010 19:05:25 +0000 (19:05 +0000)]
Merged revisions 280448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines
fixes issue with translator frame not getting freed
A translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed up.
(closes issue #17630)
Reported by: manvirr
Patches:
encoder_fix.diff uploaded by dvossel (license 671)
Tested by: manvirr, dvossel
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Jean Galarneau [Thu, 29 Jul 2010 16:01:35 +0000 (16:01 +0000)]
Merged revisions 280341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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Matthew Nicholson [Thu, 29 Jul 2010 13:45:11 +0000 (13:45 +0000)]
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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Jason Parker [Wed, 28 Jul 2010 20:02:27 +0000 (20:02 +0000)]
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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Richard Mudgett [Wed, 28 Jul 2010 19:57:49 +0000 (19:57 +0000)]
Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280229
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Jason Parker [Wed, 28 Jul 2010 19:54:54 +0000 (19:54 +0000)]
Add sha1sum-sh in case there is no util on the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280227
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Jason Parker [Wed, 28 Jul 2010 18:05:54 +0000 (18:05 +0000)]
Remove unnecessary subshells. Attempt to make checksumming work.
Also improves readability.
(issue #17621)
Reported by: bjm
Review: https://reviewboard.asterisk.org/r/808/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280193
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Sean Bright [Wed, 28 Jul 2010 16:51:11 +0000 (16:51 +0000)]
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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Leif Madsen [Wed, 28 Jul 2010 13:51:16 +0000 (13:51 +0000)]
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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David Vossel [Tue, 27 Jul 2010 20:54:32 +0000 (20:54 +0000)]
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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Jason Parker [Tue, 27 Jul 2010 17:54:54 +0000 (17:54 +0000)]
Add SHA1SUM to configure, since we require it for sounds/
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Jason Parker [Tue, 27 Jul 2010 16:48:41 +0000 (16:48 +0000)]
Remove aptly-named EMPTY and BS vars, since they aren't used anymore.
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Jason Parker [Tue, 27 Jul 2010 16:39:16 +0000 (16:39 +0000)]
Simply sounds/Makefile some more.
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Mark Michelson [Tue, 27 Jul 2010 15:13:24 +0000 (15:13 +0000)]
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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Jason Parker [Mon, 26 Jul 2010 22:59:52 +0000 (22:59 +0000)]
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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Tilghman Lesher [Mon, 26 Jul 2010 21:18:17 +0000 (21:18 +0000)]
Dunno why this worked on my machine, but it works better this way.
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Gavin Henry [Mon, 26 Jul 2010 20:25:54 +0000 (20:25 +0000)]
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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Tilghman Lesher [Mon, 26 Jul 2010 19:15:59 +0000 (19:15 +0000)]
Use a special Makefile for noobs who still have GNU Make 3.80.
(Closes issue #17716)
Reported by: farisraouf
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Sean Bright [Mon, 26 Jul 2010 15:41:13 +0000 (15:41 +0000)]
Expand the correct value within AST_OPTION_ONLY.
(closes issue #17703)
Reported by: stuarth
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Bradley Latus [Sat, 24 Jul 2010 23:58:19 +0000 (23:58 +0000)]
Minor update to man page
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Jeff Peeler [Sat, 24 Jul 2010 23:29:44 +0000 (23:29 +0000)]
Blocked revisions 279344 via svnmerge
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r279344 | jpeeler | 2010-07-24 18:27:22 -0500 (Sat, 24 Jul 2010) | 4 lines
Provide a default value for DAHDI_TRANSCODE so when DAHDI is not installed
menuselect doesn't get confused:
Unknown value '' found in build_tools/menuselect-deps for DAHDI_TRANSCODE
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Richard Mudgett [Fri, 23 Jul 2010 22:11:23 +0000 (22:11 +0000)]
Merged revisions 279206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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Mark Michelson [Fri, 23 Jul 2010 18:29:15 +0000 (18:29 +0000)]
Backport sip_uri_params_cmp() fix from trunk to 1.6.2.
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Russell Bryant [Fri, 23 Jul 2010 18:22:34 +0000 (18:22 +0000)]
remove old properties
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Russell Bryant [Fri, 23 Jul 2010 18:20:02 +0000 (18:20 +0000)]
Add branch-1.4-merged and branch-1.4-blocked properties to 1.6.2 branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279072
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Tilghman Lesher [Fri, 23 Jul 2010 17:06:17 +0000 (17:06 +0000)]
Merged revisions 278985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010) | 12 lines
Merged revisions 278984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010) | 5 lines
Establish a maximum version for openh323 (i.e. not opal), because chan_h323 will fail to load, even if it links.
(issue #17679)
Reported by: am
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Tilghman Lesher [Fri, 23 Jul 2010 16:44:53 +0000 (16:44 +0000)]
Merged revisions 278982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010) | 15 lines
Merged revisions 278981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines
Avoid race with consolethread on shutdown (on parallel processors).
(closes issue #17080)
Reported by: sybasesql
Patches:
20100721__issue17080.diff.txt uploaded by tilghman (license 14)
Tested by: sybasesql
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Tzafrir Cohen [Fri, 23 Jul 2010 15:23:09 +0000 (15:23 +0000)]
Two more typos to cancell.
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Jeff Peeler [Thu, 22 Jul 2010 19:52:17 +0000 (19:52 +0000)]
Merged revisions 278708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278708 | jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
Add method for finding XML doc files for systems that don't support GLOB_BRACE.
In particular, Solaris and perhaps others do not support the above mentioned
GNU extension. In this case the paths are simply expanded without the braces
and the calls to glob are made separately.
Note: I could not explain memory allocation failures that were being reported
from within libxml itself when making calls to glob without using GLOB_NOCHECK.
This is the only reason why that flag is being used.
(closes issue #15402)
Reported by: snuffy
Patches:
bug_xmlpatt-v3.diff uploaded by snuffy (license 35),
modified by me
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Richard Mudgett [Thu, 22 Jul 2010 19:32:16 +0000 (19:32 +0000)]
DNID does not get cleard on a new call when using immediate=yes with ISDN signaling.
When you are using chan_dahdi ISDN signaling with immediate=yes and a call
comes in without a DNID then you get the DNID of a previous call.
Chan_dahdi does not touch the DNID field on a new call if it does not have
a DNID.
Made always copy the DNID from the new call.
The patches backport the relevant changes from trunk -r210387.
(closes issue #17568)
Reported by: wuwu
Patches:
issue17568_v1.4.patch uploaded by rmudgett (license 664)
issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
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Mark Michelson [Thu, 22 Jul 2010 15:00:11 +0000 (15:00 +0000)]
Merged revisions 278620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul 2010) | 19 lines
Merged revisions 278618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines
Allow PLC to function properly when channels use SLIN for audio.
If a channel involved in a bridge was using SLIN audio, then translation
paths were not guaranteed to be set up properly since in all likelihood
the number of translation steps was only 1.
This patch enforces the transcode_via_slin behavior if transcode_via_slin
or generic_plc is enabled and one of the formats to make compatible is
SLIN.
AST-352
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Tzafrir Cohen [Wed, 21 Jul 2010 18:22:25 +0000 (18:22 +0000)]
Fix invalid test for rxisoffhook in FXO channels
This fixes some cases of no outgoing calls on FXO before an incoming call.
Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO
(KS/GS) channels.In some cases the bit would not be initialized properly
before the first inbound call and thus prevent an outgoing call.
If those tests are actually required by anybody, they should define
DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c .
(closes issue #14577)
Reported by: jkroon
Patches:
asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd (license 610)
Tested by: frawd
Review: https://reviewboard.asterisk.org/r/699/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@278524
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Russell Bryant [Wed, 21 Jul 2010 16:20:18 +0000 (16:20 +0000)]
Merged revisions 278465 via svnmerge from
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r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines
Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
This code did not properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes the limitation
on the maximum fd value.
(closes issue #15915)
Reported by: keiron
(closes issue #17187)
Reported by: Eddie Edwards
(closes issue #16494)
Reported by: Hubguru
(closes issue #15731)
Reported by: flop
(closes issue #12917)
Reported by: falves11
(closes issue #14920)
Reported by: vrban
(closes issue #17199)
Reported by: aleksey2000
(closes issue #15406)
Reported by: kowalma
(closes issue #17438)
Reported by: dcabot
(closes issue #17325)
Reported by: glwgoes
(closes issue #17118)
Reported by: erikje
possibly other issues, too ...
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Tilghman Lesher [Wed, 21 Jul 2010 15:58:20 +0000 (15:58 +0000)]
Merged revisions 278463 via svnmerge from
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r278463 | tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 lines
Ensure realtime conferences are treated the same as static conferences when trying to find an empty one.
Also, parse the useropts properly, when retrieving from realtime, and add them
to the existing flags.
(closes issue #17502)
Reported by: kenji
Patches:
20100720__issue17502.diff.txt uploaded by tilghman (license 14)
Tested by: kenji
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Tilghman Lesher [Tue, 20 Jul 2010 22:43:51 +0000 (22:43 +0000)]
Merged revisions 278275 via svnmerge from
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r278275 | tilghman | 2010-07-20 17:40:19 -0500 (Tue, 20 Jul 2010) | 14 lines
Merged revisions 278261 via svnmerge from
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r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines
Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message.
(closes issue #16350)
Reported by: noahisaac
Patches:
20100623__issue16350.diff.txt uploaded by tilghman (license 14)
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Tilghman Lesher [Tue, 20 Jul 2010 22:30:46 +0000 (22:30 +0000)]
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r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines
Merged revisions 278167 via svnmerge from
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r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110)
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Tilghman Lesher [Tue, 20 Jul 2010 16:54:20 +0000 (16:54 +0000)]
Merged revisions 278024 via svnmerge from
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r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010) | 14 lines
Merged revisions 278023 via svnmerge from
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r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines
Off-by-one error
(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
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Jean Galarneau [Mon, 19 Jul 2010 21:21:20 +0000 (21:21 +0000)]
Merged revisions 277945 via svnmerge from
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r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) | 15 lines
Merged revisions 277906 via svnmerge from
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r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines
Avoid trying to pickup a parked extension before the park operation is completed.
A crash could occur if the extension is picked up while the parking extension is
being announced. Testing pu->notquiteyet while searching for a parked extension
resolves this crash.
(ABE-2418)
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Tilghman Lesher [Sat, 17 Jul 2010 17:52:28 +0000 (17:52 +0000)]
Merge issues...
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Tilghman Lesher [Sat, 17 Jul 2010 17:43:50 +0000 (17:43 +0000)]
Merged revisions 277775 via svnmerge from
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r277775 | tilghman | 2010-07-17 12:42:32 -0500 (Sat, 17 Jul 2010) | 12 lines
Merged revisions 277738 via svnmerge from
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r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines
Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not.
(closes issue #17616)
Reported by: pprindeville
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Tilghman Lesher [Sat, 17 Jul 2010 17:41:01 +0000 (17:41 +0000)]
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r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010) | 15 lines
Merged revisions 277568 via svnmerge from
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r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines
Since we split values at the semicolon, we should store values with a semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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Tim Ringenbach [Fri, 16 Jul 2010 23:37:28 +0000 (23:37 +0000)]
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r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul 2010) | 16 lines
Merged revisions 277625 via svnmerge from
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r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines
Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer.
ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended
transfer, ast_bridge_call() is called for a second bridge on the same channel,
and it clears that flag, which still needs to get set for when the original
ast_bridge_call() gets control back and checks it.
Review: https://reviewboard.asterisk.org/r/741
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Matthew Nicholson [Fri, 16 Jul 2010 21:31:49 +0000 (21:31 +0000)]
Merged revisions 277530 via svnmerge from
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r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul 2010) | 11 lines
Merged revisions 277497 via svnmerge from
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r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128
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Jeff Peeler [Fri, 16 Jul 2010 21:16:51 +0000 (21:16 +0000)]
Merged revisions 277488 via svnmerge from
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r277488 | jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
Fix reporting estimated queue hold time.
Just say the number of seconds (after minutes) rather than doing some incorrect
calculation with respect to minutes.
(closes issue #17498)
Reported by: corruptor
Patches:
holdesecs_bug.diff uploaded by corruptor (license 253)
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Richard Mudgett [Fri, 16 Jul 2010 20:35:34 +0000 (20:35 +0000)]
Merged revisions 277467 via svnmerge from
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r277467 | rmudgett | 2010-07-16 15:27:51 -0500 (Fri, 16 Jul 2010) | 22 lines
Merged revisions 277419 via svnmerge from
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r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines
priexclusive in chan_dahdi.conf ignored when reloading dahdi module
During a reload, the priexclusive and outsignalling parameters are not
read in from the config file as intended. Unfortunately, they get set to
defaults as a result. This patch makes sure that they do not get set to
defaults during a reload.
(closes issue #17441)
Reported by: mtryfoss
Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
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Tilghman Lesher [Fri, 16 Jul 2010 20:30:40 +0000 (20:30 +0000)]
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r277452 | tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
Add documentation for MOH realtime fields
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Jeff Peeler [Fri, 16 Jul 2010 19:24:38 +0000 (19:24 +0000)]
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r277366 | jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
Add missing handling for ringing state for use with queue empty options.
(closes issue #17471)
Reported by: jazzy
Patches:
app_queue.c.diff uploaded by jazzy (license 1056)
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Matthew Nicholson [Fri, 16 Jul 2010 18:33:05 +0000 (18:33 +0000)]
Merged revisions 277331 via svnmerge from
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r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul 2010) | 15 lines
Merged revisions 277327 via svnmerge from
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r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines
Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
(closes issue #16035)
Reported by: francesco_r
Patches:
pbx.c.patch uploaded by viniciusfontes (license 978)
Tested by: francesco_r, agx, lawbar
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Tilghman Lesher [Fri, 16 Jul 2010 18:14:56 +0000 (18:14 +0000)]
Merged revisions 277263 via svnmerge from
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r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010) | 12 lines
Merged revisions 277261 via svnmerge from
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r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines
If variable gotten is not set, will segfault on Solaris.
(closes issue #17636)
Reported by: bklang
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Matthew Nicholson [Fri, 16 Jul 2010 17:31:30 +0000 (17:31 +0000)]
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r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul 2010) | 11 lines
Merged revisions 277247 via svnmerge from
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r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines
For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation.
AST-362
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Paul Belanger [Fri, 16 Jul 2010 17:18:01 +0000 (17:18 +0000)]
Merged revisions 277183 via svnmerge from
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r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul 2010) | 15 lines
Merged revisions 277182 via svnmerge from
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r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines
Total analysis time error with SIP and silence suppression
When using app_amd with SIP providers that have silence
suppression on, the iTotalTime count increases exponentially.
(closes issue #17656)
Reported by: juls
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Sean Bright [Fri, 16 Jul 2010 15:21:46 +0000 (15:21 +0000)]
Merged revisions 277143 via svnmerge from
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r277143 | seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8 lines
Avoid crashing when installing a duplicate translation path with a lower cost.
(closes issue #17092)
Reported by: moy
Patches:
translate.rev254273.patch uploaded by moy (license 222)
Tested by: moy
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Jeff Peeler [Thu, 15 Jul 2010 20:42:43 +0000 (20:42 +0000)]
Merged revisions 276788 via svnmerge from
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r276788 | jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines
Correct not setting the bindport before attempting to open the socket.
Related to changes from 276571, I was accidentally testing with a port set in
my configuration causing me to miss this. Also moved the TCP handling as well
to occur before build_peer is called.
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Jeff Peeler [Thu, 15 Jul 2010 13:52:32 +0000 (13:52 +0000)]
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r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010) | 9 lines
Merged revisions 276652 via svnmerge from
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r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines
In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is.
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Jeff Peeler [Wed, 14 Jul 2010 23:07:50 +0000 (23:07 +0000)]
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r276571 | jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines
Fix MWI notification transmission problems over SIP.
MWI updates were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status configured
previously from another mailbox.
Upon module or sip reload, MWI updates could not be sent due to the sipsock
socket not being set early enough in reload_config. The code handling the
descriptor assignment and such has simply been moved before the call to
build_peer.
Issuing a sip reload cleared the IP address of the peer, but skipped checking
the database for registration information. The database is now checked both
for sip reload and actually reloading the module.
If a transmission occurs before the do_monitor thread has started, do not
attempt to send a signal to it.
(closes issue #17398)
Reported by: ip-rob
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Kevin P. Fleming [Wed, 14 Jul 2010 20:16:28 +0000 (20:16 +0000)]
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r276441 | kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4 lines
Don't try to call an embedded module's backup_globals() function until
after confirming it exists.
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Leif Madsen [Wed, 14 Jul 2010 11:52:24 +0000 (11:52 +0000)]
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r276268 | lmadsen | 2010-07-14 06:51:48 -0500 (Wed, 14 Jul 2010) | 9 lines
Merged revisions 276267 via svnmerge from
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r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line
Update documentation for voicemail.conf externpass option.
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Russell Bryant [Tue, 13 Jul 2010 19:11:24 +0000 (19:11 +0000)]
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r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010) | 9 lines
Merged revisions 276123 via svnmerge from
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r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines
Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit).
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Jeff Peeler [Tue, 13 Jul 2010 19:01:29 +0000 (19:01 +0000)]
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r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010) | 19 lines
Merged revisions 275773 via svnmerge from
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r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines
Make user removals and traversals thread safe in meetme.
Race conditions present in meetme involving the user list where a lack of
locking has the potential for a user to be removed during a traversal or as in
the case of the reporter after checking if the list is empty could cause a
crash. Fixing this was done by convering the userlist to an ao2 container.
(closes issue #17390)
Reported by: Vince
Review: https://reviewboard.asterisk.org/r/746/
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Russell Bryant [Tue, 13 Jul 2010 16:55:22 +0000 (16:55 +0000)]
Merged revisions 275995 via svnmerge from
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r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010) | 21 lines
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r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines
Access peer->cdr directly instead of through a saved off reference.
At this point in the code, it is possible that peer_cdr may be invalid.
Specifically, in the blind transfer code, CDRs are swapped between channels.
So, peer_cdr is no longer == peer->cdr.
The scenario that exposed a crash in this code was a blind transfer that hit
the system call limit, causing the transferee channel to get destroyed after
the transfer attempt failed. Even if it succeeds and this code doesn't crash,
this code was still trying to reset a CDR on a channel that was now owned by
a different thread, which is a BadThing(tm).
(ABE-2417)
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Tilghman Lesher [Tue, 13 Jul 2010 14:49:46 +0000 (14:49 +0000)]
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r275910 | tilghman | 2010-07-13 09:48:40 -0500 (Tue, 13 Jul 2010) | 9 lines
Merged revisions 275909 via svnmerge from
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r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines
Move SQL scripts into their own database-specific directories.
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Jeff Peeler [Mon, 12 Jul 2010 17:26:10 +0000 (17:26 +0000)]
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r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010) | 18 lines
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r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines
Change ast_write to not stop generator when called from ast_prod.
For SIP channels configured with the progressinband option on, the ringback was
being immediately stopped. This problem was due to ast_prod being moved for a
deadlock fix in 259858. Prodding the channel after setting up the generator
triggered the check in ast_write to stop the generator. The fix here should
write the frame the same as was done before the call to ast_prod was moved.
(closes issue #17372)
Reported by: tech_admin
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Leif Madsen [Mon, 12 Jul 2010 15:38:11 +0000 (15:38 +0000)]
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r275626 | lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines
cdr_pgsql does not detect when a table is found.
This change adds an ERROR message to let you know when a failure exists to
get the columns from the pgsql database, which typically means that the
table does not exist.
(closes issue #17478)
Reported by: kobaz
Patches:
cdr_pgsql.patch uploaded by kobaz (license 834)
Tested by: kobaz, russell, lmadsen
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Russell Bryant [Sat, 10 Jul 2010 15:11:01 +0000 (15:11 +0000)]
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r245192 | mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21 lines
Remove useless sip options related to hash table size.
First off, these options weren't actually doing anything.
By the time the options were parsed, the peer and dialog
containers had already been allocated with their default
values.
Second, hash table size is something that doesn't really
make sense to change in a config file. If a user is that
interested in changing the hashtable size, he can modify
the source itself.
I have removed the parsing of the hash_peer, hash_user,
and hash_dialog options. I have removed the hash_user_size
variable altogether since it is not used at all. I also
changed hash_peer_size and hash_dialog_size to be constant,
and have changed the symbols to be in all caps as constants
typically are. I have also removed the entire section in
sip.conf.sample regarding configurable hashtable sizes.
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(merge to 1.6.2 inspired by issue #17553)
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Russell Bryant [Sat, 10 Jul 2010 15:08:22 +0000 (15:08 +0000)]
unblock a rev
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Russell Bryant [Fri, 9 Jul 2010 22:01:01 +0000 (22:01 +0000)]
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r275424 | russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines
Fix some issues related to dynamic feature groups in features.conf.
The bridge handling code did not properly consider feature groups when setting
parameters that would affect whether or not a native bridge would be attempted.
If DYNAMIC_FEATURES only include a feature group, a native bridge would occur
that may prevent features from working.
Fix a bug in verbose output that would show the key mapping as empty if it was
using the default mapping and not a custom mapping in the feature group.
Add feature groups to the output of "features show".
Adjust the feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group.
Update features.conf.sample to show that an '=' is still required if using
the default key mapping from [applicationmap].
Finally, clean up a little bit of formatting to better coform to coding
guidelines while in the area.
(closes issue #17589)
Reported by: lmadsen
Patches:
issue_17589.rev4.txt uploaded by russell (license 2)
Tested by: russell, lmadsen
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Russell Bryant [Fri, 9 Jul 2010 19:59:08 +0000 (19:59 +0000)]
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r275310 | russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines
Add missing ao2_iterator_destroy().
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Paul Belanger [Fri, 9 Jul 2010 19:23:19 +0000 (19:23 +0000)]
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r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul 2010) | 15 lines
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r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines
Fix logging message for stale nonce.
(closes issue #17582)
Reported by: kenner
Patches:
chan_sip.c.diff uploaded by kenner (license 1040)
Tested by: lmadsen
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Matthew Nicholson [Fri, 9 Jul 2010 18:24:45 +0000 (18:24 +0000)]
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r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul 2010) | 9 lines
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r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul 2010) | 2 lines
give a better error message when attempting to unload a module that is not loaded
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Russell Bryant [Fri, 9 Jul 2010 18:11:57 +0000 (18:11 +0000)]
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r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09 Jul 2010) | 2 lines
Move parking lot sample config out from the middle of dynamic features sample config.
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Matthew Nicholson [Fri, 9 Jul 2010 17:51:16 +0000 (17:51 +0000)]
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r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul 2010) | 9 lines
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r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines
don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded
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Matthew Nicholson [Fri, 9 Jul 2010 16:06:43 +0000 (16:06 +0000)]
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r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul 2010) | 15 lines
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r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines
Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(closes issue #17592)
Reported by: jamicque
Patches:
G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: jamicque, mnicholson
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Russell Bryant [Fri, 9 Jul 2010 15:39:22 +0000 (15:39 +0000)]
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r275022 | russell | 2010-07-09 10:35:53 -0500 (Fri, 09 Jul 2010) | 11 lines
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r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines
Document that a leading and trailing slash is expected for test categories.
Also, emit a warning if a test is registered without one of these.
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Richard Mudgett [Wed, 7 Jul 2010 18:34:27 +0000 (18:34 +0000)]
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r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010) | 1 line
Add missing conditional around chan_dahdi mfcr2_skip_category config parameter.
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Richard Mudgett [Wed, 7 Jul 2010 18:26:42 +0000 (18:26 +0000)]
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r274595 | rmudgett | 2010-07-07 13:20:00 -0500 (Wed, 07 Jul 2010) | 9 lines
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r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line
Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter.
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Tilghman Lesher [Wed, 7 Jul 2010 06:16:33 +0000 (06:16 +0000)]
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r274418 | tilghman | 2010-07-07 01:15:43 -0500 (Wed, 07 Jul 2010) | 15 lines
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r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines
Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(closes issue #16102)
Reported by: Delvar
Patches:
say.conf.fix.patch uploaded by Delvar (license 908)
(plus a few additional fixes and simplifications by me)
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Terry Wilson [Tue, 6 Jul 2010 23:06:32 +0000 (23:06 +0000)]
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r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) | 18 lines
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r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.
Review: https://reviewboard.asterisk.org/r/764/
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Jeff Peeler [Tue, 6 Jul 2010 22:30:06 +0000 (22:30 +0000)]
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r274316 | jpeeler | 2010-07-06 17:23:35 -0500 (Tue, 06 Jul 2010) | 14 lines
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r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
Correct sip.conf.sample comments for prematuremedia option.
(closes issue #17513)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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Tilghman Lesher [Tue, 6 Jul 2010 22:10:23 +0000 (22:10 +0000)]
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r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010) | 2 lines
Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield.
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Mark Michelson [Tue, 6 Jul 2010 14:33:16 +0000 (14:33 +0000)]
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r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul 2010) | 22 lines
Merged revisions 274157 via svnmerge from
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r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
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Paul Belanger [Mon, 5 Jul 2010 13:55:17 +0000 (13:55 +0000)]
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r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul 2010) | 15 lines
Merged revisions 273884 via svnmerge from
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r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines
Remove extra line breaks from 'core show config mappings'
(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
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Tilghman Lesher [Sat, 3 Jul 2010 02:43:06 +0000 (02:43 +0000)]
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r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) | 16 lines
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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Tilghman Lesher [Fri, 2 Jul 2010 17:11:38 +0000 (17:11 +0000)]
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r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010) | 15 lines
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r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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Tilghman Lesher [Fri, 2 Jul 2010 16:59:11 +0000 (16:59 +0000)]
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r273714 | tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines
The switch fallthrough could create some errorneous situations, so best to force directly to the default case.
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Tzafrir Cohen [Fri, 2 Jul 2010 15:59:30 +0000 (15:59 +0000)]
Fix typos reported by Lintian
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Russell Bryant [Thu, 1 Jul 2010 22:17:57 +0000 (22:17 +0000)]
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r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010) | 14 lines
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r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
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Jeff Peeler [Thu, 1 Jul 2010 20:29:49 +0000 (20:29 +0000)]
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r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010) | 21 lines
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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Jeff Peeler [Thu, 1 Jul 2010 15:14:41 +0000 (15:14 +0000)]
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r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010) | 19 lines
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r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@273356
65c4cc65 -6c06-0410-ace0-
fbb531ad65f3