[asterisk-commits] may: branch may/ooh323_qsig r337967 - in /team/may/ooh323_qsig: ./ apps/ chan...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Sep 25 15:39:06 CDT 2011
Author: may
Date: Sun Sep 25 15:38:38 2011
New Revision: 337967
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=337967
Log:
Merged revisions 336660-336662,336732,336735,336790,336837,336879,336937,336988,337009,337063,337117,337121-337124,337179,337220,337262,337283,337343,337346,337385,337432,337488,337543,337600,337641,337681,337722,337776,337855,337910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r336660 | rmudgett | 2011-09-19 22:57:50 +0400 (Mon, 19 Sep 2011) | 1 line
Restore 10 branch merge properties.
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r336661 | rmudgett | 2011-09-19 23:00:16 +0400 (Mon, 19 Sep 2011) | 1 line
Update merge 10 branch merge propterty.
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r336662 | rmudgett | 2011-09-19 23:03:38 +0400 (Mon, 19 Sep 2011) | 45 lines
Merged revisions 336659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r336732 | jrose | 2011-09-20 00:23:29 +0400 (Tue, 20 Sep 2011) | 21 lines
Merged revisions 336717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r336735 | tilghman | 2011-09-20 00:31:09 +0400 (Tue, 20 Sep 2011) | 25 lines
Merged revisions 336734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r336790 | tilghman | 2011-09-20 01:42:11 +0400 (Tue, 20 Sep 2011) | 9 lines
Merged revisions 336789 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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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r336837 | twilson | 2011-09-20 02:28:17 +0400 (Tue, 20 Sep 2011) | 18 lines
Merged revisions 336792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r336879 | russell | 2011-09-20 05:11:18 +0400 (Tue, 20 Sep 2011) | 50 lines
Merged revisions 336878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r336937 | irroot | 2011-09-20 20:56:11 +0400 (Tue, 20 Sep 2011) | 20 lines
Merged revisions 336936 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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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r336988 | rmudgett | 2011-09-20 22:20:10 +0400 (Tue, 20 Sep 2011) | 35 lines
Merged revisions 336978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r337009 | rmudgett | 2011-09-20 23:13:36 +0400 (Tue, 20 Sep 2011) | 29 lines
Merged revisions 337008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r337063 | kmoore | 2011-09-21 01:05:42 +0400 (Wed, 21 Sep 2011) | 25 lines
Merged revisions 337062 via svnmerge from
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r337117 | lmadsen | 2011-09-21 02:29:24 +0400 (Wed, 21 Sep 2011) | 15 lines
Merged revisions 337115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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r337121 | rmudgett | 2011-09-21 02:51:41 +0400 (Wed, 21 Sep 2011) | 1 line
Restore branch-10 merge properties.
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r337122 | rmudgett | 2011-09-21 02:53:12 +0400 (Wed, 21 Sep 2011) | 1 line
Updated 10 merge property.
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r337123 | rmudgett | 2011-09-21 02:54:21 +0400 (Wed, 21 Sep 2011) | 23 lines
Merged revisions 337119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011) | 16 lines
Fix crash with STRREPLACE function.
The ast_func_read() function calls the .read2 callback with the len
parameter set to zero indicating no size restrictions on the supplied
ast_str buffer. The value was used to dimension a local starts[] array
with the array subsequently used.
* Reworked the strreplace() function to perform the string replacement in
a straight forward manner. Eliminated the need for the starts[] array.
(closes issue ASTERISK-18545)
Reported by: Federico Alves
Patches:
jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Federico Alves
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r337124 | mjordan | 2011-09-21 03:02:25 +0400 (Wed, 21 Sep 2011) | 35 lines
Merged revisions 337120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines
Merged revisions 337118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines
Fix for incorrect voicemail duration in external notifications
This patch fixes an issue where the voicemail duration was being reported
with a duration significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the duration of
only the sound in the file, as opposed to the duration of the file with
the silence. This patch fixes this by having two durations reported in
the __ast_play_and_record family of functions - the sound_duration and the
actual duration of the file. The sound_duration, which is optional, now
reports the duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration checking, while
reporting the full duration to external parties if the voicemail is kept.
(issue ASTERISK-2234)
(closes issue ASTERISK-16981)
Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443
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r337179 | oej | 2011-09-21 13:06:22 +0400 (Wed, 21 Sep 2011) | 21 lines
Merged revisions 337178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines
Change strictrtp option to default to yes in the RTP module
Suggested by Kapejod on Facebook
Review: https://reviewboard.asterisk.org/r/1448/
(closes issue ASTERISK-18587)
Thanks for quick feedback to kpfleming and Tilghman
--Denna och nedanst?\195?\165ende rader kommer inte med i loggmeddelandet--
M CHANGES
M configs/rtp.conf.sample
M res/res_rtp_asterisk.c
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r337220 | oej | 2011-09-21 13:39:13 +0400 (Wed, 21 Sep 2011) | 22 lines
Merged revisions 337219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines
Make ast_pbx_run() not default to s at default if extension is not found
Review: https://reviewboard.asterisk.org/r/1446/
This is a bug - or architecture mistake - that has been in Asterisk for a
very long time. It was exposed by the AMI originate action and possibly
some other applications. Most channel drivers checks if an extension
exists BEFORE starting a pbx on an inbound call, so most calls will
not depend on this issue.
Thanks everyone involved in the review and on IRC and the mailing list
for a quick review and all the feedback.
(closes issue ASTERISK-18578)
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r337262 | irroot | 2011-09-21 14:46:09 +0400 (Wed, 21 Sep 2011) | 16 lines
Merged revisions 337261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21 Sep 2011) | 10 lines
Adds a timeout argument to app_originate
the default is 30s this will be used if the timout supplied is invalid or
no timeout is supplied.
Contributed by: jacco (thank you for the work)
Review: https://reviewboard.asterisk.org/r/1310/
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r337283 | irroot | 2011-09-21 15:21:49 +0400 (Wed, 21 Sep 2011) | 9 lines
Merged revisions 337263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) | 1 line
Whitespace fixup from SRTP patch
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r337343 | tilghman | 2011-09-22 00:53:13 +0400 (Thu, 22 Sep 2011) | 12 lines
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Escape commas in keys and values, when keys and values are enumerated by commas.
Review: https://reviewboard.asterisk.org/r/1433
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Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8
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Merged revisions 337342 from https://origsvn.digium.com/svn/asterisk/branches/10
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r337346 | tilghman | 2011-09-22 01:10:14 +0400 (Thu, 22 Sep 2011) | 10 lines
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Dumb little spacing fix.
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Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 337345 from http://svn.asterisk.org/svn/asterisk/branches/10
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r337385 | tilghman | 2011-09-22 01:26:34 +0400 (Thu, 22 Sep 2011) | 12 lines
More silly spacing changes
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Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 337380 from http://svn.asterisk.org/svn/asterisk/branches/10
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r337432 | irroot | 2011-09-22 10:39:01 +0400 (Thu, 22 Sep 2011) | 32 lines
Merged revisions 337431 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337431 | irroot | 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) | 25 lines
Merged revisions 337430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines
Its possible to loose audio on ast_write when the channel is not transcoded correctly.
in the case of DAHDI the channel is hungup.
This patch tries to "fix" the problem and make the channel compatiable and warn the user of
this problem.
Please note there is a underlying problem with codec negotion this does not fix the problem
it does try to rectify it and prevent loss of service.
Review: https://reviewboard.asterisk.org/r/1442/
(closes issue ASTERISK-17541)
(closes issue ASTERISK-18063)
(issue ASTERISK-14384)
(issue ASTERISK-17502)
(issue ASTERISK-18325)
(issue ASTERISK-18422)
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r337488 | irroot | 2011-09-22 13:31:41 +0400 (Thu, 22 Sep 2011) | 23 lines
Merged revisions 337487 via svnmerge from
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r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines
Merged revisions 337486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines
If IP address is used in chan_h323 host parameter of peer configuration.
module tries to resolve IP address to IP address and fails.
Simple fix to set family of socket this is a hangover from ipv6 changes.
(closes issue ASTERISK-18237)
(issue ASTERISK-17278)
(issue ASTERISK-17500)
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r337543 | irroot | 2011-09-22 15:46:35 +0400 (Thu, 22 Sep 2011) | 21 lines
Merged revisions 337542 via svnmerge from
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r337542 | irroot | 2011-09-22 13:44:22 +0200 (Thu, 22 Sep 2011) | 14 lines
Merged revisions 337541 via svnmerge from
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r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines
Add warned to ast_srtp to prevent errors on each frame from libsrtp
The first 9 frames are not reported as some devices dont use srtp
from first frame these are suppresed.
the warning is then output only once every 100 frames.
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r337600 | jrose | 2011-09-22 20:35:20 +0400 (Thu, 22 Sep 2011) | 30 lines
Merged revisions 337595,337597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines
Generate Security events in chan_sip using new Security Events Framework
Security Events Framework was added in 1.8 and support was added for AMI to generate
events at that time. This patch adds support for chan_sip to generate security events.
(closes issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
Review: https://reviewboard.asterisk.org/r/1362/
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r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines
Forgot to svn add new files to r337595
Part of Generating security events for chan_sip
(issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
Reviewboard: https://reviewboard.asterisk.org/r/1362/
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r337641 | pabelanger | 2011-09-22 22:44:26 +0400 (Thu, 22 Sep 2011) | 11 lines
Blocked revisions 337640 via svnmerge
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r337640 | pabelanger | 2011-09-22 14:43:35 -0400 (Thu, 22 Sep 2011) | 5 lines
Revert previous commit
New feature should be added into trunk, unfortunately it is too late for the
Asterisk 10 branch.
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r337681 | irroot | 2011-09-23 00:03:33 +0400 (Fri, 23 Sep 2011) | 17 lines
Blocked revisions 337433 via svnmerge
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r337433 | irroot | 2011-09-22 08:42:42 +0200 (Thu, 22 Sep 2011) | 12 lines
Revert commit r337261
This commit is for trunk not version 10
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Adds a timeout argument to app_originate
the default is 30s this will be used if the timout supplied is invalid or
no timeout is supplied.
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r337722 | rmudgett | 2011-09-23 01:42:35 +0400 (Fri, 23 Sep 2011) | 32 lines
Merged revisions 337721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines
Merged revisions 337720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines
Made ISDN not add numbering plan prefix strings to empty numbers.
When the Caller-ID is restricted, the expected behavior is for the
Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto
the Caller-ID number even if it is restricted (empty) causing the
Caller-ID to be the national prefix rather than blank.
This behavior was lost when sig_pri was extracted from chan_dahdi.
* Made not add prefix strings to empty connected line, calling, and ANI
number strings.
(closes issue ASTERISK-18577)
Reported by: Kris Shaw
Patches:
jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Kris Shaw
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r337776 | russell | 2011-09-23 04:47:18 +0400 (Fri, 23 Sep 2011) | 25 lines
Merged revisions 337775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r337775 | russell | 2011-09-22 19:45:35 -0500 (Thu, 22 Sep 2011) | 18 lines
Merged revisions 337774 via svnmerge from
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r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) | 11 lines
Comment out entries in sample res_pktccops.conf.
With these options enabled, they can cause Asterisk to freak out by
SYN flooding a network and eating the CPU. Obviously it would be good to
fix the code so that this can't happen, but we can at least change the default
configuration so it doesn't happen.
This was reported downstream to the Fedora issue tracker:
https://bugzilla.redhat.com/show_bug.cgi?id=658431
........
................
................
r337855 | irroot | 2011-09-23 13:35:32 +0400 (Fri, 23 Sep 2011) | 24 lines
Merged revisions 337840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................
r337840 | irroot | 2011-09-23 10:39:22 +0200 (Fri, 23 Sep 2011) | 17 lines
Merged revisions 337839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines
Make sure a CDR is on the stack for call in the Queue.
Only let update_cdr act on the last CDR in the stack.
In some circumstances [Attended transfer to queue] a
CDR record is not inserted for this call where it should.
(closes issue ASTERISK-18567)
Review: https://reviewboard.asterisk.org/r/1266
........
................
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r337910 | irroot | 2011-09-23 23:20:41 +0400 (Fri, 23 Sep 2011) | 17 lines
Merged revisions 337902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................
r337902 | irroot | 2011-09-23 21:18:14 +0200 (Fri, 23 Sep 2011) | 10 lines
Merged revisions 337898 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | 4 lines
Spelling fix
........
................
................
Added:
team/may/ooh323_qsig/channels/sip/include/security_events.h
- copied unchanged from r337910, trunk/channels/sip/include/security_events.h
team/may/ooh323_qsig/channels/sip/security_events.c
- copied unchanged from r337910, trunk/channels/sip/security_events.c
Modified:
team/may/ooh323_qsig/ (props changed)
team/may/ooh323_qsig/CHANGES
team/may/ooh323_qsig/Makefile
team/may/ooh323_qsig/Makefile.rules
team/may/ooh323_qsig/UPGRADE-1.8.txt
team/may/ooh323_qsig/apps/app_confbridge.c
team/may/ooh323_qsig/apps/app_dial.c
team/may/ooh323_qsig/apps/app_echo.c
team/may/ooh323_qsig/apps/app_followme.c
team/may/ooh323_qsig/apps/app_meetme.c
team/may/ooh323_qsig/apps/app_minivm.c
team/may/ooh323_qsig/apps/app_mixmonitor.c
team/may/ooh323_qsig/apps/app_morsecode.c
team/may/ooh323_qsig/apps/app_mp3.c
team/may/ooh323_qsig/apps/app_originate.c
team/may/ooh323_qsig/apps/app_queue.c
team/may/ooh323_qsig/apps/app_saycounted.c
team/may/ooh323_qsig/apps/app_voicemail.c
team/may/ooh323_qsig/channels/chan_h323.c
team/may/ooh323_qsig/channels/chan_sip.c
team/may/ooh323_qsig/channels/sig_pri.c
team/may/ooh323_qsig/channels/sig_ss7.c
team/may/ooh323_qsig/channels/sip/include/sdp_crypto.h
team/may/ooh323_qsig/channels/sip/include/sip.h
team/may/ooh323_qsig/channels/sip/include/srtp.h
team/may/ooh323_qsig/channels/sip/sdp_crypto.c
team/may/ooh323_qsig/codecs/gsm/Makefile
team/may/ooh323_qsig/configs/extensions.conf.sample
team/may/ooh323_qsig/configs/logger.conf.sample
team/may/ooh323_qsig/configs/res_pktccops.conf.sample
team/may/ooh323_qsig/configs/rtp.conf.sample
team/may/ooh323_qsig/configs/sip.conf.sample
team/may/ooh323_qsig/configure
team/may/ooh323_qsig/configure.ac
team/may/ooh323_qsig/contrib/init.d/rc.archlinux.asterisk
team/may/ooh323_qsig/contrib/init.d/rc.redhat.asterisk
team/may/ooh323_qsig/contrib/scripts/get_ilbc_source.sh
team/may/ooh323_qsig/funcs/func_curl.c
team/may/ooh323_qsig/funcs/func_strings.c
team/may/ooh323_qsig/include/asterisk/app.h
team/may/ooh323_qsig/include/asterisk/autoconfig.h.in
team/may/ooh323_qsig/include/asterisk/event_defs.h
team/may/ooh323_qsig/include/asterisk/optional_api.h
team/may/ooh323_qsig/include/asterisk/security_events_defs.h
team/may/ooh323_qsig/main/Makefile
team/may/ooh323_qsig/main/app.c
team/may/ooh323_qsig/main/channel.c
team/may/ooh323_qsig/main/event.c
team/may/ooh323_qsig/main/pbx.c
team/may/ooh323_qsig/main/security_events.c
team/may/ooh323_qsig/res/res_musiconhold.c
team/may/ooh323_qsig/res/res_rtp_asterisk.c
team/may/ooh323_qsig/res/res_srtp.c
team/may/ooh323_qsig/tests/test_pbx.c
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- branch-10-blocked (original)
+++ branch-10-blocked Sun Sep 25 15:38:38 2011
@@ -1,1 +1,1 @@
-/branches/10:330492,330514
+/branches/10:330492,330514,337433,337640
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Sep 25 15:38:38 2011
@@ -1,1 +1,1 @@
-/trunk:1-336529,336571,336600
+/trunk:1-337966
Modified: team/may/ooh323_qsig/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/CHANGES?view=diff&rev=337967&r1=337966&r2=337967
==============================================================================
--- team/may/ooh323_qsig/CHANGES (original)
+++ team/may/ooh323_qsig/CHANGES Sun Sep 25 15:38:38 2011
@@ -213,6 +213,8 @@
SIP Changes
-----------
* Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
+ * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
+ * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
Queue changes
-------------
@@ -232,6 +234,7 @@
a MeetMe conference
* Added 'k' option to MeetMe to automatically kill the conference when there's only
one participant left (much like a normal call bridge)
+ * Added extra argument to Originate to set timeout.
Asterisk Database
-----------------
@@ -248,8 +251,28 @@
IAX2 Changes
------------
-* authdebug is now disabled by default. To enable this functionaility again
+ * authdebug is now disabled by default. To enable this functionaility again
set authdebug = yes in iax.conf.
+
+RTP Changes
+-----------
+ * The rtp.conf setting "strictrtp" is now enabled by default. In previous
+ releases it was disabled.
+
+PBX Core
+--------
+ * The PBX core previously made a call with a non-existing extension test for
+ extension s at default and jump there if the extension existed.
+ This was a bad default behaviour and violated the principle of least surprise.
+ It has therefore been changed in this release. It may affect some
+ applications and configurations that rely on this behaviour. Most channel
+ drivers have avoided this for many releases by testing whether the extension
+ called exists before starting the PBX and generating a local error.
+ This behaviour still exists and works as before.
+
+ Extension "s" is used when no extension is given in a channel driver,
+ like immediate answer in DAHDI or calling to a domain with no user part
+ in a SIP uri.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
Modified: team/may/ooh323_qsig/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/Makefile?view=diff&rev=337967&r1=337966&r2=337967
==============================================================================
--- team/may/ooh323_qsig/Makefile (original)
+++ team/may/ooh323_qsig/Makefile Sun Sep 25 15:38:38 2011
@@ -286,7 +286,7 @@
ifneq ($(findstring darwin,$(OSARCH)),)
_ASTCFLAGS+=-D__Darwin__
SOLINK=-bundle -Xlinker -macosx_version_min -Xlinker 10.4 -Xlinker -undefined -Xlinker dynamic_lookup -force_flat_namespace
- ifeq ($(shell /usr/bin/sw_vers -productVersion | cut -c1-4),10.6)
+ ifeq ($(shell if test `/usr/bin/sw_vers -productVersion | cut -c4` -gt 5; then echo 6; else echo 0; fi),6)
SOLINK+=/usr/lib/bundle1.o
endif
_ASTLDFLAGS+=-L/usr/local/lib
Modified: team/may/ooh323_qsig/Makefile.rules
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/Makefile.rules?view=diff&rev=337967&r1=337966&r2=337967
==============================================================================
--- team/may/ooh323_qsig/Makefile.rules (original)
+++ team/may/ooh323_qsig/Makefile.rules Sun Sep 25 15:38:38 2011
@@ -37,8 +37,8 @@
OPTIMIZE?=-O6
ifneq ($(findstring darwin,$(OSARCH)),)
- ifeq ($(shell /usr/bin/sw_vers -productVersion | cut -c1-4),10.6)
- # Snow Leopard has an issue with this optimization flag on large files (like chan_sip)
+ ifeq ($(shell if test `/usr/bin/sw_vers -productVersion | cut -c4` -gt 5; then echo 6; else echo 0; fi),6)
+ # Snow Leopard/Lion has an issue with this optimization flag on large files (like chan_sip)
OPTIMIZE+=-fno-inline-functions
endif
endif
Modified: team/may/ooh323_qsig/UPGRADE-1.8.txt
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/UPGRADE-1.8.txt?view=diff&rev=337967&r1=337966&r2=337967
==============================================================================
--- team/may/ooh323_qsig/UPGRADE-1.8.txt (original)
+++ team/may/ooh323_qsig/UPGRADE-1.8.txt Sun Sep 25 15:38:38 2011
@@ -143,6 +143,12 @@
events/responses output the connected line ID as caller ID. These party ID's
are now separate.
+* The Dial application d and H options do not automatically answer the call
+ anymore. It broke DTMF attended transfers. Since many SIP and ISDN phones
+ cannot send DTMF before a call is connected, you need to answer the call
+ leg to those phones before using Dial with these options for them to have
+ any effect before the dialed party answers.
+
* The outgoing directory (where .call files are read) now uses inotify to
detect file changes instead of polling the directory on a regular basis.
If your outgoing folder is on a NFS mount or another network file system,
Modified: team/may/ooh323_qsig/apps/app_confbridge.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_confbridge.c?view=diff&rev=337967&r1=337966&r2=337967
==============================================================================
--- team/may/ooh323_qsig/apps/app_confbridge.c (original)
+++ team/may/ooh323_qsig/apps/app_confbridge.c Sun Sep 25 15:38:38 2011
@@ -1283,6 +1283,7 @@
10,
"sln",
&duration,
+ NULL,
ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE),
0,
NULL);
Modified: team/may/ooh323_qsig/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_dial.c?view=diff&rev=337967&r1=337966&r2=337967
==============================================================================
--- team/may/ooh323_qsig/apps/app_dial.c (original)
+++ team/may/ooh323_qsig/apps/app_dial.c Sun Sep 25 15:38:38 2011
@@ -120,6 +120,11 @@
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
if it exists.</para>
+ <note>
+ <para>Many SIP and ISDN phones cannot send DTMF digits until the call is
+ connected. If you wish to use this option with these phones, you
+ can use the <literal>Answer</literal> application before dialing.</para>
+ </note>
</option>
<option name="D" argsep=":">
<argument name="called" />
@@ -170,10 +175,18 @@
</note>
</option>
<option name="h">
- <para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
+ <para>Allow the called party to hang up by sending the DTMF sequence
+ defined for disconnect in <filename>features.conf</filename>.</para>
</option>
<option name="H">
- <para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
+ <para>Allow the calling party to hang up by sending the DTMF sequence
+ defined for disconnect in <filename>features.conf</filename>.</para>
+ <note>
+ <para>Many SIP and ISDN phones cannot send DTMF digits until the call is
+ connected. If you wish to allow DTMF disconnect before the dialed
+ party answers with these phones, you can use the <literal>Answer</literal>
+ application before dialing.</para>
+ </note>
</option>
<option name="i">
<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
@@ -1759,7 +1772,7 @@
*/
silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
ast_answer(chan);
- res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
+ res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, NULL, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of
conflicts by naming the pa.privintro file */
if (res == -1) {
@@ -2070,10 +2083,6 @@
res = -1; /* reset default */
}
- if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
[... 5009 lines stripped ...]
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