David Vossel [Fri, 6 Nov 2009 15:42:46 +0000 (15:42 +0000)]
Merged revisions 228338 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines
fixes crash in astfd.c
(closes issue #15981)
Reported by: slavon
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228339
65c4cc65-6c06-0410-ace0-
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David Vossel [Fri, 6 Nov 2009 15:08:04 +0000 (15:08 +0000)]
Blocked revisions 228272 via svnmerge
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r228272 | dvossel | 2009-11-06 09:07:31 -0600 (Fri, 06 Nov 2009) | 9 lines
fixes memory leak in func_audiohookinherit.c
(closes issue 0015394)
Reported by: boroda
Patches:
bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
Tested by: dbrooks, boroda
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228273
65c4cc65-6c06-0410-ace0-
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David Vossel [Fri, 6 Nov 2009 15:04:24 +0000 (15:04 +0000)]
fixes memory leak in func_audiohookinherit.c
(closes issue #15394)
Reported by: boroda
Patches:
bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
Tested by: dbrooks, boroda
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228268
65c4cc65-6c06-0410-ace0-
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Mark Michelson [Thu, 5 Nov 2009 22:59:02 +0000 (22:59 +0000)]
Fix XML in func_cdr.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228233
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Thu, 5 Nov 2009 22:12:45 +0000 (22:12 +0000)]
Yet another error message in the dialplan (thanks, rmudgett/russellb)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228196
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Thu, 5 Nov 2009 21:24:21 +0000 (21:24 +0000)]
MEETME_INFO should not return a literal error message to the dialplan.
(closes issue #15450)
Reported by: JimVanM
Patches:
meetmeinfopatch.diff.txt uploaded by dbrooks (license 790)
Tested by: JimVanM
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228191
65c4cc65-6c06-0410-ace0-
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Jeff Peeler [Thu, 5 Nov 2009 21:23:06 +0000 (21:23 +0000)]
Fix the fix for chanspy option o
In 224178, I assumed the uploaded patch was correct as it had received positive
feedback. The flags were being checked in the incorrect location. Upon testing
the fix this time it was also found that the flags from the dialplan weren't
being copied to the chanspy_translation_helper.
(closes issue #16167)
Reported by: marhbere
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228189
65c4cc65-6c06-0410-ace0-
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David Brooks [Thu, 5 Nov 2009 19:34:50 +0000 (19:34 +0000)]
Merged revisions 228078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines
chan_misdn Asterisk 1.4.27-rc2 crash
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested
by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached
a full bt." This patch zeros out an ast_frame.
(closes issue #16041)
Reported by: francesco_r
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228145
65c4cc65-6c06-0410-ace0-
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Jason Parker [Thu, 5 Nov 2009 19:16:29 +0000 (19:16 +0000)]
Merged revisions 228079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines
Fix crash on VPB exception when no hardware is present.
(closes issue #14970)
Reported by: tzafrir
Patches:
vpb_exception.diff uploaded by tzafrir (license 46)
Tested by: markwaters
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228080
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Thu, 5 Nov 2009 17:26:55 +0000 (17:26 +0000)]
Rework codecs command to comply with the 64-bit scheme
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228049
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Thu, 5 Nov 2009 17:08:02 +0000 (17:08 +0000)]
Don't crash if no arguments are passed.
(closes issue #16119)
Reported by: thedavidfactor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228015
65c4cc65-6c06-0410-ace0-
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Jeff Peeler [Wed, 4 Nov 2009 23:50:59 +0000 (23:50 +0000)]
Merged revisions 227944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines
Fix incorrect filename comparsion after monitor file change
The logic to detect if a requested file is indeed a different file from the
current file was incorrect. The main issue being confusion of the use of
filename_base which was previously set without pathing information and then
compared to another full path. Robust file comparison logic has been added
to properly check if two files are the same even if symlinks are used.
(closes issue #15313)
Reported by: caspy
Patches:
20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325)
but mostly tilghman's work
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227945
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Jeff Peeler [Wed, 4 Nov 2009 22:22:51 +0000 (22:22 +0000)]
Update chan_ooh323 to support the expanded codec bitfield from 227580.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227914
65c4cc65-6c06-0410-ace0-
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Alexandr Anikin [Wed, 4 Nov 2009 22:10:44 +0000 (22:10 +0000)]
Reworked chan_ooh323 channel module.
Many architectural and functional changes.
Main changes are threading model chanes (many thread in ooh323 stack
instead of one), modifications and improvements in signalling part,
additional codecs support (726, speex), t38 mode support.
This module tested and used in production environment.
(closes issue #15285)
Reported by: may213
Tested by: sles, c0w, OrNix
Review: https://reviewboard.asterisk.org/r/324/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227898
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Wed, 4 Nov 2009 21:39:33 +0000 (21:39 +0000)]
Added the 'a' option to app dial and modified app_dial to set the answertime when the called channel answers.
This change causes answertime to be correct even if the called channel hangs up during an announcement triggered by the A() option.
(closes issue #15936)
Reported by: falves11
Patches:
dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96)
dial-caller-answer1.diff uploaded by mnicholson (license 96)
Tested by: falves11, mnicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227897
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Wed, 4 Nov 2009 21:03:33 +0000 (21:03 +0000)]
Merged revisions 227827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines
This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements.
(closes issue #16005)
Reported by: falves11
Patches:
dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, falves11
Review: https://reviewboard.asterisk.org/r/407/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227829
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Tilghman Lesher [Wed, 4 Nov 2009 20:35:24 +0000 (20:35 +0000)]
Fixes for gcc 4.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227824
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Wed, 4 Nov 2009 20:13:50 +0000 (20:13 +0000)]
Modify the SDP parsing code to parse session and media level items separately.
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future.
(closes issue #14994)
Reported by: frawd
Tested by: frawd, mnicholson, file
Review: https://reviewboard.asterisk.org/r/414/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227759
65c4cc65-6c06-0410-ace0-
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Joshua Colp [Wed, 4 Nov 2009 19:26:19 +0000 (19:26 +0000)]
Merged revisions 227735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines
Fix a security issue where it may be possible for someone to execute a cross-site
AJAX request exploit.
(AST-2009-009)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227739
65c4cc65-6c06-0410-ace0-
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Joshua Colp [Wed, 4 Nov 2009 19:20:46 +0000 (19:20 +0000)]
Merged revisions 227700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines
Fix a security issue where sending a REGISTER with a differing username in the From
URI and Authorization header would reveal whether it was valid or not.
(AST-2009-008)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227712
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Mark Michelson [Wed, 4 Nov 2009 16:41:02 +0000 (16:41 +0000)]
Add a couple more casts so that code compiles correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227646
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 4 Nov 2009 16:35:27 +0000 (16:35 +0000)]
mmichelson reported a compilation error related to codec bit expansion that should be resolved with a simple include of frame_defs.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227645
65c4cc65-6c06-0410-ace0-
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Jeff Peeler [Wed, 4 Nov 2009 16:25:15 +0000 (16:25 +0000)]
fix trunk building
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227643
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 4 Nov 2009 16:17:18 +0000 (16:17 +0000)]
Two other trunk build fixes (reported by seanbright on #asterisk-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227615
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 4 Nov 2009 16:13:50 +0000 (16:13 +0000)]
Fix trunk building
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227614
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 4 Nov 2009 14:05:12 +0000 (14:05 +0000)]
Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 4 Nov 2009 13:57:09 +0000 (13:57 +0000)]
chan_misdn will fail to compile if the redirect_dn member is missing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227579
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Olle Johansson [Wed, 4 Nov 2009 08:22:00 +0000 (08:22 +0000)]
Add destruction of iterators to avoid problems with refcounters
(per Russell's review of another patch)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227545
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 4 Nov 2009 03:15:10 +0000 (03:15 +0000)]
Don't crash when state_interface is NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227509
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Tue, 3 Nov 2009 22:13:25 +0000 (22:13 +0000)]
Resolve another warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227464
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Tue, 3 Nov 2009 22:08:46 +0000 (22:08 +0000)]
Resolve a warning from gcc 4.4.1.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227463
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Tue, 3 Nov 2009 22:05:31 +0000 (22:05 +0000)]
Resolve some dev-mode warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227462
65c4cc65-6c06-0410-ace0-
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David Brooks [Tue, 3 Nov 2009 21:26:28 +0000 (21:26 +0000)]
AMI hook interface
This patch, originally submitted by jozza, enables custom modules to send actions to AMI
and receive messages from AMI via a hook interface. Included is a simple test module to
illustrate the interface.
(closes issue #14635)
Reported by: jozza
Review: https://reviewboard.asterisk.org/r/412/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227448
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Tue, 3 Nov 2009 21:21:09 +0000 (21:21 +0000)]
This patch adds a sequence field to CDRs that can be combined with the linkedid or uniqueid field to uniquely identify a CDR.
(closes issue #15180)
Reported by: Nick_Lewis
Patches:
cdr-sequence10.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435
65c4cc65-6c06-0410-ace0-
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Joshua Colp [Tue, 3 Nov 2009 21:16:14 +0000 (21:16 +0000)]
Add support for using a hint when configuring a state interface using the format hint:<extension>@<context>.
(closes issue #15168)
Reported by: p_lindheimer
Patches:
queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227424
65c4cc65-6c06-0410-ace0-
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Jason Parker [Tue, 3 Nov 2009 19:59:46 +0000 (19:59 +0000)]
Fix some build issues on Solaris.
(closes issue #14517)
(SWP-109)
Reported by: asgaroth
Patches:
bug_14517.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy, dougm, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227372
65c4cc65-6c06-0410-ace0-
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Leif Madsen [Tue, 3 Nov 2009 19:48:53 +0000 (19:48 +0000)]
Change warning message to debug message.
app_controlplayback outputs a warning, when in fact it is normal.
(closes issue #16071)
Reported by: atis
Patches:
controlplayback_warning.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227368
65c4cc65-6c06-0410-ace0-
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Leif Madsen [Tue, 3 Nov 2009 19:25:18 +0000 (19:25 +0000)]
Additional fixes to the extensions.conf.sample file.
Update the extensions.conf.sample [stdexten] context so that we use the
variable instead of requiring it to be passed explicitly. Also updated uses of
the [stdexten] context throughout.
(closes issue #15858)
Reported by: pprindeville
Patches:
stdexten-context-update.txt uploaded by lmadsen (license 10)
Tested by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227361
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Tue, 3 Nov 2009 18:22:28 +0000 (18:22 +0000)]
Fixed a spelling error in the q850 reason header option in the output of sip show settings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227298
65c4cc65-6c06-0410-ace0-
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Richard Mudgett [Tue, 3 Nov 2009 17:58:38 +0000 (17:58 +0000)]
Recorded merge of revisions 227275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines
Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls.
This is the relevant portion of asterisk/trunk -r226648
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227277
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Tilghman Lesher [Tue, 3 Nov 2009 17:56:41 +0000 (17:56 +0000)]
Code guidelines fixes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227276
65c4cc65-6c06-0410-ace0-
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David Vossel [Tue, 3 Nov 2009 17:12:52 +0000 (17:12 +0000)]
user.conf entries in SIP were not having their peer type set.
(closes issue #16120)
Reported by: jsmith
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227238
65c4cc65-6c06-0410-ace0-
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Olle Johansson [Tue, 3 Nov 2009 16:56:48 +0000 (16:56 +0000)]
Adding some clarifications to func_speex doxygen docs.
The functions needed doesn't exist in Speex 1.05 which is what a lot of distros use.
1.2 seems to have been in beta status for years, and does include the sexy functions needed for func_speex to work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227237
65c4cc65-6c06-0410-ace0-
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Joshua Colp [Tue, 3 Nov 2009 15:37:08 +0000 (15:37 +0000)]
Merged revisions 227166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines
Fix a bug where an RPID header could be generated with a blank username in the URI.
(closes issue #15909)
Reported by: kobaz
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227167
65c4cc65-6c06-0410-ace0-
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Leif Madsen [Tue, 3 Nov 2009 15:19:47 +0000 (15:19 +0000)]
Update extensions.conf.sample file to fix incorrect extensions.
(closes issue #15857)
Reported by: pprindeville
Patches:
stdexten.patch#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227162
65c4cc65-6c06-0410-ace0-
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Olle Johansson [Tue, 3 Nov 2009 11:11:15 +0000 (11:11 +0000)]
Merged revisions 227088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines
Use proper response code when violating Contact ACL's.
https://reviewboard.asterisk.org/r/415/
Thanks kpfleming for a quick review.
(EDVX-003)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227091
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Tilghman Lesher [Mon, 2 Nov 2009 22:29:19 +0000 (22:29 +0000)]
Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks
(closes issue #12950)
Reported by: alea-soluciones
Patches:
ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514)
Tested by: alea-soluciones, adomjan, urtho, nahuelgreco
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227049
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David Brooks [Mon, 2 Nov 2009 20:59:37 +0000 (20:59 +0000)]
SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226974
65c4cc65-6c06-0410-ace0-
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David Brooks [Mon, 2 Nov 2009 20:57:45 +0000 (20:57 +0000)]
SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226973
65c4cc65-6c06-0410-ace0-
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Olle Johansson [Mon, 2 Nov 2009 20:43:52 +0000 (20:43 +0000)]
Adding external reference for doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226970
65c4cc65-6c06-0410-ace0-
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Joshua Colp [Mon, 2 Nov 2009 18:08:54 +0000 (18:08 +0000)]
Merged revisions 226889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
while the called party had not yet answered.
This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
(closes issue #14674)
Reported by: ulogic
Patches:
bug14674.patch uploaded by jpeeler (license 325)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226890
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Richard Mudgett [Mon, 2 Nov 2009 17:34:22 +0000 (17:34 +0000)]
DAHDI ISDN channel names will not allow device state to work. (Interim solution.)
Since ISDN works like SIP and not analog ports in regard to devices, the
device state based on the ISDN channel number could not work. This has
not been an issue until the advent of PTMP NT mode. Previously, ISDN
lines were used as trunks and did not have to keep track of specific
devices.
As an interim solution until device states are properly implemented, the
channel name is being changed to the following format to use the generic
device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
Dialplan hints would thus be:
exten => xxx,hint,DAHDI/i2/5551212
This will work with the following restrictions:
* The number of devices/phones cannot exceed the number of B channels.
(i.e., BRI has 2)
* Each device/phone can only have one number. No shared MSN's.
* The phones/devices probably should not use subaddressing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882
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Tilghman Lesher [Mon, 2 Nov 2009 17:15:31 +0000 (17:15 +0000)]
Merged revisions 226811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines
Don't allow two separate instances of safe_asterisk when restarting from the init script.
(closes issue #14562)
Reported by: davidw
Patches:
Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780)
Tested by: davidw
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226812
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David Vossel [Mon, 2 Nov 2009 15:34:37 +0000 (15:34 +0000)]
Blocked revisions 226736 via svnmerge
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r226736 | dvossel | 2009-11-02 09:31:02 -0600 (Mon, 02 Nov 2009) | 5 lines
fixes crash on iterator_destroy on uninitialized iterator
(closes issue #16162)
Reported by: krn
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226748
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David Vossel [Mon, 2 Nov 2009 15:17:04 +0000 (15:17 +0000)]
Blocked revisions 226688 via svnmerge
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r226688 | dvossel | 2009-11-02 09:16:30 -0600 (Mon, 02 Nov 2009) | 5 lines
changes calltoken debug messages from LOG_NOTICE to LOG_DEBUG like they are supposed to be
(closes issue #16144)
Reported by: aragon
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226689
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Matthew Nicholson [Mon, 2 Nov 2009 14:57:11 +0000 (14:57 +0000)]
This patch adds support for a draft proposal for adding Q.850 reason headers to sip messages.
(closes issue #13385)
Reported by: adomjan
Patches:
sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487)
CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487)
chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487)
sip-q850-hangupcause1.diff uploaded by mnicholson (license 96)
Tested by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687
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Richard Mudgett [Fri, 30 Oct 2009 23:26:41 +0000 (23:26 +0000)]
Cleanup some flags on DAHDI PRI channel hangup.
* Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split)
* Make sure the outgoing flag is cleared if a new channel fails to get
created for outgoing calls.
* Remove some unused flags since sig_pri was split.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226648
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Russell Bryant [Fri, 30 Oct 2009 04:08:39 +0000 (04:08 +0000)]
Add an "Asterisk Architecture Overview" section to the doxygen documentation.
This is a side project I've been poking at this week. The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together. There is a ton of stuff to write about, so this will
just continue to evolve over time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606
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Joshua Colp [Thu, 29 Oct 2009 18:13:42 +0000 (18:13 +0000)]
Merged revisions 226531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines
Add an option to enabling passing music on hold start and stop requests through instead of
acting on them in chan_local.
(closes issue #14709)
Reported by: dimas
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532
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Olle Johansson [Thu, 29 Oct 2009 12:20:16 +0000 (12:20 +0000)]
Doxygen documentation update
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226490
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Tzafrir Cohen [Wed, 28 Oct 2009 20:50:52 +0000 (20:50 +0000)]
remove empty awk pattern (//)
Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'.
Just remove that. No pattern at all always matches.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226453
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Leif Madsen [Wed, 28 Oct 2009 20:11:07 +0000 (20:11 +0000)]
Merged revisions 226382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines
Update documentation in sip.conf.sample.
Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.
(closes issue #15644)
Reported by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226384
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Leif Madsen [Wed, 28 Oct 2009 19:50:00 +0000 (19:50 +0000)]
Merged revisions 226377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines
Update CALLINGSUBADDR channel variable documentation.
(closes issue #15734)
Reported by: alecdavis
Patches:
channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226378
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Tilghman Lesher [Wed, 28 Oct 2009 18:04:05 +0000 (18:04 +0000)]
Merged revisions 226304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines
Fix documentation (pointed out by TheDavidFactor on #-dev)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226305
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Tzafrir Cohen [Wed, 28 Oct 2009 08:47:59 +0000 (08:47 +0000)]
Remove extra cleanup in case we have more than one Asterisk.
/var/run would be cleaned on startup on most systems anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226270
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Tzafrir Cohen [Tue, 27 Oct 2009 22:10:38 +0000 (22:10 +0000)]
another variation of the upstart script
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226227
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Olle Johansson [Tue, 27 Oct 2009 21:03:22 +0000 (21:03 +0000)]
Adding compile time flags for Snow Leopard, Leopard and some other animals
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226184
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Tilghman Lesher [Tue, 27 Oct 2009 20:22:07 +0000 (20:22 +0000)]
Merged revisions 226138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines
Manager output is not always NULL-terminated, so force a NULL at the end of the filestream.
(closes issue #15495)
Reported by: pdf
Patches:
20090916__issue15495.diff.txt uploaded by tilghman (license 14)
Tested by: pdf
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226159
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Terry Wilson [Tue, 27 Oct 2009 16:48:54 +0000 (16:48 +0000)]
Don't prepend the URI prefix to the post directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226099
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Joshua Colp [Tue, 27 Oct 2009 13:30:27 +0000 (13:30 +0000)]
Add support for receiving unsolicited MWI NOTIFY messages.
This change adds a configuration option to SIP peers, unsolicited_mailbox, which
configures a virtual mailbox to use for received new/old MWI information. This
virtual mailbox can then be used by any device supporting MWI.
(closes issue #13028)
Reported by: AsteriskRocks
Patches:
bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060
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Tzafrir Cohen [Mon, 26 Oct 2009 22:46:09 +0000 (22:46 +0000)]
detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi
* When checking if we are Linux, check OSARCH rather than host_os
The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than
'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case.
OSARCH is tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH rather
than $host_os .
See also: http://wiki.debian.org/ArmEabiPort
Merged revisions 225957 via svnmerge from
http://svn.digium.com/svn/asterisk/branches/1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226018
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Kevin P. Fleming [Mon, 26 Oct 2009 22:04:04 +0000 (22:04 +0000)]
Fix building in REF_DEBUG mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225956
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Kevin P. Fleming [Mon, 26 Oct 2009 22:03:29 +0000 (22:03 +0000)]
Correct broken logic from revision 225405.
The code committed in revision 225405 was broken; instead of removing the unreference code,
the logic used to decide when to do it should have been reversed. This patch corrects the
situation, and makes reference counting work properly again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225955
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Jeff Peeler [Mon, 26 Oct 2009 19:40:26 +0000 (19:40 +0000)]
ACL check not present for verifying SIP INVITEs
The ACL check in check_peer_ok was missing and has now been restored. The
missing check allowed for calls to be made on prohibited networks where an ACL
was defined in sip.conf and the allowguest option was set to off. See the AST
security advisory below for more information.
Merge code associated with AST-2009-007.
(closes issue #16091)
Reported by: thom4fun
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225912
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Richard Mudgett [Mon, 26 Oct 2009 16:07:09 +0000 (16:07 +0000)]
Make conditionals create previous code when libpri/ss7 are present.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225872
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Tzafrir Cohen [Mon, 26 Oct 2009 13:29:54 +0000 (13:29 +0000)]
span numbers in pri debug / error messages
Prefix PRI trace messages with the span number. This makes the trace
readable even when you have a multi-port device.
(closes issue #15054)
Reported by: tzafrir
Patches:
dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225836
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Tzafrir Cohen [Mon, 26 Oct 2009 11:34:06 +0000 (11:34 +0000)]
Re-arange code a bit to build in dev-mode without ss7
No change of functionality here. Just localized a variable and indented
code into blocks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225803
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Tzafrir Cohen [Mon, 26 Oct 2009 09:40:49 +0000 (09:40 +0000)]
Make chan_dahdi build even without PRI / SS7
(Note: still some strange build warnings without SS7 in dev-mode)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225767
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Kevin P. Fleming [Sat, 24 Oct 2009 14:40:37 +0000 (14:40 +0000)]
Improve performance of pedantic mode dialog searching in chan_sip.
This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support
to make pedantic mode dialog searching in find_call() not require a linear search
of all dialogs in the list of dialogs. This patch does *not* change the dialog
matching logic (more on that later), just improves the searching performance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727
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Richard Mudgett [Fri, 23 Oct 2009 16:57:33 +0000 (16:57 +0000)]
Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692
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Sean Bright [Fri, 23 Oct 2009 16:40:30 +0000 (16:40 +0000)]
Optionally build and install the sample AGIs in the agi/ directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225690
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David Vossel [Fri, 23 Oct 2009 14:41:50 +0000 (14:41 +0000)]
Fixes an iterator memory leak and uninitialized memory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225650
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Kevin P. Fleming [Fri, 23 Oct 2009 14:02:42 +0000 (14:02 +0000)]
Merged revisions 225581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines
Don't force menuselect.makeopts to be rebuilt on every build.
For some reason the menuselect.makeopts file was listed as PHONY in the Makefile,
resulting in 'make' needing to rebuild it for every build. This then resulted in
the embedded module rules being rebuilt on every build, which can be slow and is
unnecessary.
This patch fixes the problem by properly allowing 'make' to know when the
menuselect.makeopts file needs to be rebuilt (defining the proper dependencies).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225582
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Leif Madsen [Thu, 22 Oct 2009 22:24:03 +0000 (22:24 +0000)]
Update README documentation.
Update the README documentation to correctly describe which CLI command you should
use when attempting to get help from the CLI.
(closes issue #16064)
Reported by: thedavidfactor
Patches:
readme.patch uploaded by thedavidfactor (license 903)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225515
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Leif Madsen [Thu, 22 Oct 2009 21:52:30 +0000 (21:52 +0000)]
Merged revisions 225484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines
Clean valgrind output by suppressing false errors.
Update valgrind.txt documentation and add valgrind.supp file in order to
allow those who are creating valgrind output to have less false errors in
the logfile.
(closes issue #16007)
Reported by: atis
Patches:
valgrind.txt.diff uploaded by atis (license 242)
asterisk2.supp uploaded by atis (license 242)
Tested by: atis, amorsen
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225485
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Leif Madsen [Thu, 22 Oct 2009 21:28:44 +0000 (21:28 +0000)]
Add Asterisk Git HowTo documentation.
Added documentation on how to create a local git repository from
SVN. This documentation was added via doxygen.
(closes issue #15814)
Reported by: tzafrir
Patches:
git-asterisk-howto uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225483
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Richard Mudgett [Thu, 22 Oct 2009 20:07:55 +0000 (20:07 +0000)]
Search for the subaddress only within the extension section of the dial string.
Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225446
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David Vossel [Thu, 22 Oct 2009 19:55:51 +0000 (19:55 +0000)]
SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445
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Sean Bright [Thu, 22 Oct 2009 19:33:32 +0000 (19:33 +0000)]
Add the programs in utils/ to menuselect.
Nothing in utils/ is now built by default except for astcanary.
Review: https://reviewboard.asterisk.org/r/353/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225440
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Tilghman Lesher [Thu, 22 Oct 2009 19:10:04 +0000 (19:10 +0000)]
Permit storage of voicemail secrets in a separate file, located within the spool directory.
(closes issue #14276)
Reported by: klaus3000
Patches:
app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65)
Tested by: jamesgolovich
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406
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Kevin P. Fleming [Thu, 22 Oct 2009 18:41:47 +0000 (18:41 +0000)]
Fix a refcount error introduced by yesterday's OBJ_MULTIPLE commit.
When an object is being unlinked from its container *and* being returned to
the caller, we do not want to decrement the reference count after unlinking
it from the container, as the reference that the container held is what we
are returning to the caller... and if it was the only remaining reference to
the object, that could result in the object being destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225405
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Tilghman Lesher [Thu, 22 Oct 2009 17:11:23 +0000 (17:11 +0000)]
Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103)
Reported by: majorbloodnok
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360
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Richard Mudgett [Thu, 22 Oct 2009 16:33:22 +0000 (16:33 +0000)]
Add support for calling and called subaddress. Partial support for COLP subaddress.
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.
(closes issue #15604)
Reported by: alecdavis
Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
Some minor modificatons were made.
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357
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David Vossel [Wed, 21 Oct 2009 21:58:46 +0000 (21:58 +0000)]
Merged revisions 225243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
IAX2: VNAK loop caused by signaling frames with no destination call number
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307
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Kevin P. Fleming [Wed, 21 Oct 2009 21:15:40 +0000 (21:15 +0000)]
Add 'mohsuggest' configuration option to 'sip show peer' CLI command and
SIPShowPeer AMI action.
(closes issue #15990)
Reported by: _brent_
Patches:
sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388)
Review: https://reviewboard.asterisk.org/r/381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245
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Kevin P. Fleming [Wed, 21 Oct 2009 21:08:47 +0000 (21:08 +0000)]
Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244
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Russell Bryant [Wed, 21 Oct 2009 16:46:22 +0000 (16:46 +0000)]
Merged revisions 225171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) | 2 lines
Revert 225169, as this doesn't account for the possibility of a list of frames.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225172
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Russell Bryant [Wed, 21 Oct 2009 16:42:13 +0000 (16:42 +0000)]
Merged revisions 225169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) | 2 lines
Isolate the frame returned from ast_translate().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225170
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Tilghman Lesher [Wed, 21 Oct 2009 15:46:42 +0000 (15:46 +0000)]
Blocked revisions 225103 via svnmerge
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r225103 | tilghman | 2009-10-21 10:45:54 -0500 (Wed, 21 Oct 2009) | 2 lines
Suffix is not needed for a match
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225104
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Tilghman Lesher [Wed, 21 Oct 2009 15:42:47 +0000 (15:42 +0000)]
Apparently, I don't need to specify the ".so" suffix to get a match
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225102
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