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r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines
Update RedHat Init script to work with Heartbeat.
The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that
it can work correctly with Heartbeat.
(Closes issue ASTERISK-18253)
Reported by: c0rnoTa
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r337062 | kmoore | 2011-09-20 16:05:01 -0500 (Tue, 20 Sep 2011) | 18 lines
Merged revisions 337061 via svnmerge from
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r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | 11 lines
Make CANMATCH with the new pattern match engine behave more like the old one
When checking an extension for E_CANMATCH using the new extension matching
algorithm, an exact match was not returned as a possible match resulting in the
queue failing to allow a caller to exit on DTMF. This removes the requirement
that an extension be longer than acquired digits for an E_CANMATCH operation
to succeed.
(closes issue ASTERISK-18044)
Review: https://reviewboard.asterisk.org/r/1367/
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r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines
Merged revisions 337007 via svnmerge from
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r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines
Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().
Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.
* Added some missing libss7 access lock protection.
* Prevent cancelling the ss7_linkset() thread at inoportune times just
like the pri_dchannel() thread.
(issue ASTERISK-17955)
Reported by: Ian M Sherman
Patches:
jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
(attached to related ASTERISK-17966)
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r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines
Merged revisions 336977 via svnmerge from
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r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines
Fix deadlock from not releasing SS7 linkset lock.
sig_ss7_hangup() failed to release the SS7 linkset lock if the call had
the alreadyhungup flag set.
* Made unlock the SS7 linkset lock in sig_ss7_hangup() if the
alreadyhungup flag is set.
* Made ss7_start_call() not hold any locks while creating the channel for
an incoming call to prevent deadlock.
* Made ss7_grab() a void function, since it could never fail, to simplify
calling code.
* Made obtain the channel lock to do softhangup in some places.
Patches:
jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett
JIRA AST-668
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r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
Allow Setting Auth Tag Bit length Based on invite or config option
Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
Curently only 80 bit is supported.
The outgoing invite will use the taglen of the incoming invite preventing
one-way audio.
(Closes issue ASTERISK-17895)
Review: https://reviewboard.asterisk.org/r/1173/
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r336878 | russell | 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) | 43 lines
Merged revisions 336877 via svnmerge from
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r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines
Fix crashes in ast_rtcp_write().
This patch addresses crashes related to RTCP handling. The backtraces just
show a crash in ast_rtcp_write() where it appears that the RTP instance is no
longer valid. There is a race condition with scheduled RTCP transmissions and
the destruction of the RTP instance. This patch utilizes the fact that
ast_rtp_instance is a reference counted object and ensures that it will not get
destroyed while a reference is still around due to scheduled RTCP
transmissions.
RTCP transmissions are scheduled and executed from the chan_sip scheduler
context. This scheduler context is processed in the SIP monitor thread. The
destruction of an RTP instance occurs when the associated sip_pvt gets
destroyed (which happens when the sip_pvt reference count reaches 0). However,
the SIP monitor thread is not the only thread that can cause a sip_pvt to get
destroyed. The sip_hangup function, executed from a channel thread, also
decrements the reference count on a sip_pvt and could cause it to get
destroyed.
While this is being changed anyway, the patch also removes calling
ast_sched_del() from within the RTCP scheduler callback. It's not helpful.
Simply returning 0 prevents the callback from being rescheduled.
(closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(issue ASTERISK-17560)
(issue ASTERISK-15406)
(issue ASTERISK-15257)
(issue ASTERISK-13334)
(issue ASTERISK-9977)
(issue ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
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r336792 | twilson | 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) | 9 lines
Merged revisions 336791 via svnmerge from
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r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines
Don't interfere with T.38 reinvites
This is an update to the fix for ASTERISK-18340 and ASTERISK-17725
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r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011) | 2 lines
Ensure substring will not be found in the previous match.
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r336734 | tilghman | 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) | 18 lines
Merged revisions 336733 via svnmerge from
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r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines
Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)
* Makefile workaround for 10.6 extended to work on 10.7 and later.
* Now uses the 'weak' symbol for Lion systems, which no longer support
'weak_import'
Closes ASTERISK-17612.
Closes ASTERISK-18213.
Tested by: tilghman, oej.
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r336717 | jrose | 2011-09-19 15:16:23 -0500 (Mon, 19 Sep 2011) | 14 lines
Merged revisions 336716 via svnmerge from
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r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines
Document applications that play audio and do not answer unanswered calls.
This patch is part of an effort to document early media and its usage. If you are
interested in contributing to this documentation effort, there are probably other
applications worth documenting as well as an Asterisk wiki article at
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
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r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines
Merged revisions 336658 via svnmerge from
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r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
Made Dial d and H options no longer immediately auto-answer the calling leg.
The Dial d and H options break DTMF attended transfer atxferdropcall
option.
1) Party A calls party B.
2) Party B does a DTMF attended transfer to Party C.
If the dialplan uses the Dial d or H options to call Party C then the Dial
application answers the call immediately before initiating the call leg to
Party C. The premature answer causes the transfer code to not invoke the
atxferdropcall=no behavior for a blonde transfer since Party C has
"answered". The transfer code thinks that Party B has "consulted" with
Party C when Party B hangs up and completes the transfer to Party A.
Party A now hears ringback until Party C actually answers.
ASTERISK-13294 Dial d option.
ASTERISK-11067 Dial H option to disconnect before answer.
The referenced issues made Dial answer with the d and H options because
many SIP and ISDN phones cannot send DTMF before the call is connected.
* Made require the dialplan to control when or if the call needs to be
answered to use the Dial application d and H options. (The call is no
longer surprise answered when using the Dial d or H options.)
Review: https://reviewboard.asterisk.org/r/1381/
JIRA AST-623
JIRA AST-666
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r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines
Update get_ilbc_source.sh script to work again.
Recently iLBC support in Asterisk has changed after the acquisition of GIPS
by Google. More information about how this may affect you is available in a
blog post at:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
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r336570 | rmudgett | 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) | 11 lines
Merged revisions 336569 via svnmerge from
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r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines
Rework sig_pri_hangup() to be simpler and clearer.
JIRA AST-675
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r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) | 12 lines
Merged revisions 336501 via svnmerge from
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r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines
Add diversion header to a 302 redirect response if we have diversion data
(closes issue ASTERISK-18143)
patch by oej
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r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines
Merged revisions 336499 via svnmerge from
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r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines
A long time ago in a galaxy far far away a IPv6 update was made,
chan_h323 was not updated causeing all to flee to chan_ooh323.
the brave Jedi [asterisk developers] pondered this miscarrige of justice
and restored order to the force for the sake of closing out 2 old issues.
(closes issue ASTERISK-17278)
(closes issue ASTERISK-17500)
Reported by: dread, sybasesql
Tested by: irroot
Reviewed by: IRC (russellb, kpfleming)
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r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån, 19 Sep 2011) | 9 lines
Merged revisions 336440 via svnmerge from
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r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines
Make sure manager_debug option is reset at reload
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r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) | 16 lines
Merged revisions 336378 via svnmerge from
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r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines
Add missing unlock at MWI message sending time
(closes issue ASTERISK-18573)
Patches:
sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
Thanks to irrot for the reminder, to Gregory for the patch!
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r336316 | twilson | 2011-09-16 17:11:39 -0500 (Fri, 16 Sep 2011) | 9 lines
Merged revisions 336314 via svnmerge from
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r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines
Whitespace fix
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r336313 | twilson | 2011-09-16 17:07:00 -0500 (Fri, 16 Sep 2011) | 12 lines
Merged revisions 336312 via svnmerge from
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r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines
Add missing frame types to func_frame_trace
Also casts control frames to the proper enum so that the compile will catch
new additions.
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r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines
Merged revisions 336294 via svnmerge from
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r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a new type of control frame
so that our RTP bridge loop can properly detect when these situations occur and check to see
if peers need to be updated in order to send their media to the proper location.
(Closes issue ASTERISK-18340)
Reported by: Thomas Arimont
(Closes issue ASTERISK-17725)
Reported by: kwk
Tested by: twilson, jrose
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r336235 | seanbright | 2011-09-16 15:10:39 -0400 (Fri, 16 Sep 2011) | 9 lines
Merged revisions 336234 via svnmerge from
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r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines
Make a note that inotify won't work with an NFS mounted spooler directory.
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r336167 | irroot | 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) | 22 lines
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r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines
The round robin routing routine in chan_misdn.c is broken.
it rotates between ports but never checks the channels in the ports.
i have extensivly tested it and verified it works on 1 upto 4 ports.
before the patch only 1 out of each port was used now all are used as
expected.
(closes issue ASTERISK-18413)
Reported by: irroot
Tested by: irroot
Reviewed by: irroot
Review: https://reviewboard.asterisk.org/r/1410/
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r336094 | irroot | 2011-09-15 17:54:46 +0200 (Thu, 15 Sep 2011) | 26 lines
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r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines
Locking order in app_queue.c causes deadlocks.
a channel lock must never be held with the queues container lock held.
the deadlock occured on masquerade.
the queues container lock is a relic of the past the old queue module lock.
with ao2 there is no need to hold this lock when dealing with members this
patch removes unneeded locks.
(closes issue ASTERISK-18101)
(closes issue ASTERISK-18487)
Reported by: Paul Rolfe, Jason Legault
Tested by: irroot, Jason Legault, Paul Rolfe
Reviewed by: Matthew Nicholson
Review: https://reviewboard.asterisk.org/r/1402/
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r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011) | 2 lines
Removes some no-op code found in format_cap.c.
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r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
When using Meetme as a modular call bridge from third party applications, it's handy to make
it behave like a normal call bridge. When the second to last person exists, the last person
will be kicked out of the conference when this option is enabled.
(closes issue ASTERISK-18234)
Review: https://reviewboard.asterisk.org/r/1376/
Patch by oej, sponsored by ClearIT, Solna, Sweden
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r335991 | irroot | 2011-09-15 10:29:12 +0200 (Thu, 15 Sep 2011) | 17 lines
Merged revisions 335978 via svnmerge from
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r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines
lock the channel before calling ast_bridged_channel() to prevent a seg fault.
AMI agents list called on shutdown causes a segfault, introducing proper locking
will prevent this.
(closes issue ASTERISK-18092)
Reported by: agustina
Patches: chan_agent.patch (License #5041) patch uploaded by irroot
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r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines
Merged revisions 335911 via svnmerge from
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r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines
Remove unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated an error
about not having PRI_L2_PERSISTENCE if you did not have the absolute
latest libpri SVN checkout installed.
The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to
be for libraries that are dependent upon other libraries and not
necessarily for optional/added features within a library.
(closes issue ASTERISK-18535)
Reported by: Michael Keuter
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r335852 | rmudgett | 2011-09-14 11:00:37 -0500 (Wed, 14 Sep 2011) | 18 lines
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r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines
Fixed cut-n-paste regression using the wrong variable.
Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
sections for channel configuration.
(closes issue ASTERISK-18496)
Reported by: Sean Darcy
Patches:
jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Sean Darcy, rmudgett
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r335791 | mnicholson | 2011-09-14 08:28:50 -0500 (Wed, 14 Sep 2011) | 11 lines
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r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines
The tech and data members of fast_originate_helper are not string fields.
ASTERISK-17709
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r335721 | rmudgett | 2011-09-13 17:10:44 -0500 (Tue, 13 Sep 2011) | 9 lines
Merged revisions 335720 via svnmerge from
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r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line
Remove obsolete todo comment about PICKUPRESULT.
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Review: https://reviewboard.asterisk.org/r/1432/
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Do parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the build-time
default language (normally "en") is always the default one.
Review: https://reviewboard.asterisk.org/r/1342/
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>
Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716
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r335656 | tilghman | 2011-09-13 13:55:33 -0500 (Tue, 13 Sep 2011) | 11 lines
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r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines
Move mandatory checks closer to the beginning of the file.
If these are going to fail, they should fail as quickly as possible.
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r335653 | mnicholson | 2011-09-13 13:47:57 -0500 (Tue, 13 Sep 2011) | 12 lines
Merged revisions 335618 via svnmerge from
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r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines
Don't limit the size of appdata for manager originate actions.
ASTERISK-17709
Patch by: tilghman (with modifications)
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Review: https://reviewboard.asterisk.org/r/1434/
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Review: https://reviewboard.asterisk.org/r/1427/
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Review: https://reviewboard.asterisk.org/r/1432/
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r335510 | russell | 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) | 22 lines
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r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines
Fix a crash in res_ais.
This patch resolves a crash observed in a load testing environment that
involved the use of the res_ais module. I observed some crashes where
the event delivery callback would get called, but the length parameter
incidcating how much data there was to read was 0. The code assumed
(with good reason I would think) that if this callback got called, there
was an event available to read. However, if the rare case that there's
nothing there, catch it and return instead of blowing up.
More specifically, the change always ensure that the size of the received
event in the cluster is always big enough to be a real ast_event.
Review: https://reviewboard.asterisk.org/r/1423/
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r335434 | mnicholson | 2011-09-12 10:55:48 -0500 (Mon, 12 Sep 2011) | 13 lines
Merged revisions 335433 via svnmerge from
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r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines
Properly set caller_warning and callee_warning before we try to use them.
ASTERISK-18199
Patch by: elguero
Testing by: rtang
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r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines
Merged revisions 335341 via svnmerge from
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r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines
Ensure frames are not written to dialed channel if ringback is requested
When a single channel was dialed and there was media to be forwarded to the
calling channel, the media was written without regard for ringback causing
silence to be heard in some circumstances. This regression was introduced
when the meaning of "single" changed to mean only the number of channels
dialed.
(closes issue ASTERISK-18083)
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devices
Review: https://reviewboard.asterisk.org/r/1429/
(closes issue ASTERISK-18497)
Thanks to russellb for peer review.
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r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines
Merged revisions 335319 via svnmerge from
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r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines
Lock the peer->mvipvt to avoid crashes with SIP history enabled
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
which cause issues with SIP history additions in combination with the max limit for
number of history entries.
Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288)
Thanks to irrot for peer review. Work with this bug funded by IPvision AS
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r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines
Merged revisions 335320 via svnmerge from
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r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines
Prevent IAX2 from getting IPv6 addresses via DNS
IAX2 does not support IPv6 and getting such addresses from DNS can cause error
messages on the remote end involving bad IPv4 address casts in the presence of
IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6
addresses via DNS queries.
(closes issue ASTERISK-18090)
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r335260 | schmidts | 2011-09-12 11:11:45 +0000 (Mon, 12 Sep 2011) | 12 lines
Merged revisions 335259 via svnmerge from
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r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines
build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value.
adding an ao2_unlink from the peers_by_ip container fix it.
Review: https://reviewboard.asterisk.org/r/1428/
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