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r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines
s/1.10/10.0/
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r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines
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r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines
Document parkinglot in chan_dahdi.conf.sample.
* Document existing feature in chan_dahdi.conf.sample.
* Remove some dead code related to the parkinglot option.
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r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines
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r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines
Update PickupChan documentation.
The PickupChan uses the ampersand as the argument separator.
Was documented as:
PickupChan(channel[,channel2[,...][,options]])
Fixed documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144)
Reported by: Erik Smith
Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
Tested by: Erik Smith
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r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines
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r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines
Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked!
This appears to be a leftover from when ast_channel was converted to ao2
objects.
Simply removed the extraneous unlock.
(closes issue ASTERISK-17772)
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r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines
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r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines
Asterisk now requires libpri 1.4.11+ for PRI support.
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r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines
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r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines
We can't guarantee an eth0 is present
FreeBSD test fails on this case presumably because there is no eth0 on the test
machine. Better to just remove this test for now.
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r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
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r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
Inband DTMF regression
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
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r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines
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r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines
Revert partial attempt at handling pathnames with spaces.
Revision 299794 attempted to improve the build system to be able to handle
pathnames (primarily DESTDIR) with spaces in them, since this is common on
some platforms (including Mac OSX). Unfortunately, the changes were incomplete
and did not actually provide the desired behavior, and as a side effect the
functionality that ensured that stale headers in the Asterisk 'include' directory
were removed got broken. In addition, the check for stale (and possibly
incompatible) modules in the Asterisk 'modules' directory also got broken, and
would never report any stale modules. Users upgrading to this version or later
versions would then see unexpected module load errors.
Since there are few users who actually want to install Asterisk into paths
that contain spaces, and a proper fix for the build system would take many hours,
the best solution for now is to just revert the partial solution.
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r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
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r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
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r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328770 via svnmerge from
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r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
MeetMe requests a PIN twice in some circumstances
If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
options, MeetMe will ask for the PIN two times: once for creating the
conference and once for entering the conference. This behavior was introduced
in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
controlling PIN entry for joining a conference.
(closes AST-601)
Review: https://reviewboard.asterisk.org/r/1305/
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r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines
Merged revisions 328716 via svnmerge from
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r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines
Make AST_LIST_REMOVE safer
AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This
commit also adds linked list unit tests.
Review: https://reviewboard.asterisk.org/r/1321/
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r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
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r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
app_dial may double free a channel datastore
When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917)
Reported by: Mark Murawski
Tested by: Mark Murawski
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r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
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r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
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r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines
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r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines
Fixed invalid read and null pointer deref on asterisk shutdown.
In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash.
(closes issue ASTERISK-17927)
Reported by: Mark Murawski
Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher
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r328541 | tilghman | 2011-07-18 02:11:26 -0500 (Mon, 18 Jul 2011) | 9 lines
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r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines
Typo
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r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines
Merged revisions 328427 via svnmerge from
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r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines
small gk processing fixes:
- decrease for 1 second registration ttl for very low expirations (some
providers expire few earlier than TTL)
- delete rrq and registration expire timers on URQ received as we make
new registration.
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r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines
delete unproperly changed svn props
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r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line
Build app_macro by default because things depend on it.
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r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines
Update UPGRADE.txt and CHANGES files.
Update documentation files stating that deprecated modules are no longer built by default.
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Adds sublines to skinny lines. Each subline can be attached to an
SLA station/trunk combo. Includes the following functionality:
Callid is persistent for both in/out calls on all skinny devices.
Can join, hold, resume.
All sublines appear under a single line button.
See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc.
(closes issue ASTERISK-17947)
Review: https://reviewboard.asterisk.org/r/1239/
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r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines
Make hint watcher callback take const strings for context and exten parameters.
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r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines
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r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines
Missing SIP pvt and channel unlock in sip_set_rtp_peer().
Regression introduced by -r326144.
Add missing SIP pvt and channel unlock in sip_set_rtp_peer().
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines
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r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines
Monitor application arguments requirements fixed.
Monitor was requiring options in spite of no individual option on Monitor being required.
Review: https://reviewboard.asterisk.org/r/1320/
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r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines
tune the v21 preamble detector to properly detect the desired sequence of hits
and misses
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r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines
Preserve sample rate quality of wideband mixmonitor recordings.
MixMonitor has the ability to record in any file format Asterisk supports,
but the quality of wideband audio is not preserved. This is because
regardless of the sample rate the call is being recorded in, the audio
is always downsampled to 8khz and then upsampled to whatever wideband
format it is being written as. This patch resolves this by requesting
the audio from the audiohook in the signed linear format closest to the
sample rate of the format we are writing. This fix is only possible for
Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
audio.
Review: https://reviewboard.asterisk.org/r/1314/
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r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) | 1 line
Add ATXFER_NULL_TECH note in features.conf.sample.
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r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines
Correct double-free situation in manager output processing.
The process_output() function calls ast_str_append() and xml_translate() on its
'out' parameter, which is a pointer to an ast_str buffer. If either of these
functions need to reallocate the ast_str so it will have more space, they will
free the existing buffer and allocate a new one, returning the address of the
new one. However, because process_output only receives a pointer to the ast_str,
not a pointer to its caller's variable holding the pointer, if the original
ast_str is freed, the caller will not know, and will continue to use it (and
later attempt to free it).
(reported by jkroon on #asterisk-dev)
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r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) | 2 lines
search in the current context for 'a' and 'o' instead of 'default'
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r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | 1 line
Fix uninstall target, so that modules dir gets cleared again.
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r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines
Added additional checks for mailbox / password beginning with '*' character
A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.
(closes issue ASTERISK-17443)
Reported by: Kevin Scott Adams
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1316/
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r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines
Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability.
The problem with using 'echo -n' is that it is not portable. While BSD systems
required that the '-n' option be removed and interpreted, System V required
that all strings should be echoed with no interpretation of options. This
fundamental difference of behavior means that it is never possible to use the
'-n' flag to echo in tests which are meant to be portable.
In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the
System V semantics of the command, and thus the SHELL test failed on that
platform.
http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
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When undergoing a shutdown and channels are kicked out of a bridge, a segfault
occurs because ConfBridge tries to play sounds on the bridge after the
underlying channels have been blown away due to the shutdown.
(closes ASTERISK-18040)
Review: https://reviewboard.asterisk.org/r/1283/
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines
Update chan_gtalk to work with changed GMail-based calls
The messages sent by the GMail client have changed, but include the
old-style messages as well. This patch checks for this case and
uses the old-style offer.
(closes issue ASTERISK-18084)
Review: https://reviewboard.asterisk.org/r/1312/
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follow_talker mode originally echoed the same video stream
to all participants. As the primary talker switched around, the
video stream would result in the talker seeing themselves. Now
the primary talker sees the last person who was talking rather than
themselves.
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul 2011) | 2 lines
reset our buffer each iteration when doing variable substitution
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