Russell Bryant [Wed, 9 Jun 2010 10:28:27 +0000 (10:28 +0000)]
Add libradiusclient-ng-dev to install_prereq.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269198
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Russell Bryant [Wed, 9 Jun 2010 10:23:05 +0000 (10:23 +0000)]
Add libbluetooth-dev to install_prereq.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269197
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Russell Bryant [Wed, 9 Jun 2010 10:21:23 +0000 (10:21 +0000)]
Add libmysqlclient-dev to install_prereq.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269196
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Russell Bryant [Wed, 9 Jun 2010 10:18:24 +0000 (10:18 +0000)]
Add libgtk2.0-dev to the packages list for install_prereq.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269187
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Bradley Latus [Tue, 8 Jun 2010 23:48:17 +0000 (23:48 +0000)]
Add High Resolution Times to CDRs for Asterisk
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153
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Tilghman Lesher [Tue, 8 Jun 2010 22:45:16 +0000 (22:45 +0000)]
Fix build on Mac OS X (and maybe FreeBSD, too)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269119
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Matthew Nicholson [Tue, 8 Jun 2010 18:50:45 +0000 (18:50 +0000)]
Don't pass null to manager_event()
(closes issue #17087)
Reported by: bklang
Patches:
app-fax-null-sprintf1.diff uploaded by mnicholson (license 96)
Tested by: bklang
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269083
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Russell Bryant [Tue, 8 Jun 2010 15:41:23 +0000 (15:41 +0000)]
Ensure CONFIG_FLAGS makes it into the build rules when doing out of tree builds.
(closes issue #16685)
Reported by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269008
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Sean Bright [Tue, 8 Jun 2010 15:39:52 +0000 (15:39 +0000)]
Merged revisions 269006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun 2010) | 11 lines
Reduce startup time for cdr_tds with large CDR tables.
Since we are just checking for table existence, add a WHERE clause that will
return no rows but will raise an error if the table doesn't exist.
(closes issue #17380)
Reported by: kkwong
Patches:
issue17380-01.patch uploaded by seanbright (license 71)
Tested by: kkwong
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269007
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Leif Madsen [Tue, 8 Jun 2010 15:23:20 +0000 (15:23 +0000)]
Update note in sip.conf.sample.
Update note in sip.conf.sample about externip and externhost with STUN.
(closes issue #16323)
Reported by: klaus3000
Patches:
sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268988
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Leif Madsen [Tue, 8 Jun 2010 14:38:18 +0000 (14:38 +0000)]
Fix some doxygen warnings.
(closes issue #17336)
Reported by: snuffy
Patches:
doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969
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Tilghman Lesher [Tue, 8 Jun 2010 06:57:24 +0000 (06:57 +0000)]
Release list lock before returning on error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268933
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Tilghman Lesher [Tue, 8 Jun 2010 06:16:43 +0000 (06:16 +0000)]
Fix trunk build on Mac OS X.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268896
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Terry Wilson [Tue, 8 Jun 2010 05:29:08 +0000 (05:29 +0000)]
Add SRTP support for Asterisk
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894
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Richard Mudgett [Tue, 8 Jun 2010 00:45:13 +0000 (00:45 +0000)]
Make SIP tests compile again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268857
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Tilghman Lesher [Mon, 7 Jun 2010 22:56:53 +0000 (22:56 +0000)]
Use the mailbox destructor function, instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268818
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Tilghman Lesher [Mon, 7 Jun 2010 22:47:13 +0000 (22:47 +0000)]
Mailbox list would previously grow at each reload, containing duplicates.
Also, optimize the allocation of mailboxes to avoid additional memory structures.
(closes issue #16320)
Reported by: Marquis
Patches:
20100525__issue16320.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268817
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Richard Mudgett [Mon, 7 Jun 2010 20:04:42 +0000 (20:04 +0000)]
Extract sig_ss7 out of chan_dahdi.
Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling. The new SS7 structures were modeled on
sig_pri.
The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.
1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().
2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.
3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.
Review: https://reviewboard.asterisk.org/r/661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774
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Tilghman Lesher [Mon, 7 Jun 2010 19:52:39 +0000 (19:52 +0000)]
Seems strange (and the code backs up) that if the max and min of a statistic is expressed as a double, the last value would not also need to be a double.
(closes issue #15807)
Reported by: klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268773
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Richard Mudgett [Mon, 7 Jun 2010 19:06:55 +0000 (19:06 +0000)]
Moved AOC request code out of the middle of code parsing the dialed number.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268734
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Tilghman Lesher [Mon, 7 Jun 2010 18:59:27 +0000 (18:59 +0000)]
Event well was going dry.
(issue #17234)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268731
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Paul Belanger [Mon, 7 Jun 2010 17:34:45 +0000 (17:34 +0000)]
Set threshold for silence detection defaults to 256
(closes issue #15685)
Reported by: david_s5
Patches:
dsp-silence-threshold-init.diff uploaded by dant (license 670)
issue15685.patch.v5 uploaded by pabelanger (license 224)
Tested by: danti
Review: https://reviewboard.asterisk.org/r/670/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268690
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Tilghman Lesher [Mon, 7 Jun 2010 17:14:40 +0000 (17:14 +0000)]
Avoid unloading res_smdi twice.
(closes issue #17237)
Reported by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268653
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Richard Mudgett [Mon, 7 Jun 2010 15:51:39 +0000 (15:51 +0000)]
Suppress warning in waitstream_core().
Suppress the warning about unexpected control subclass frames for
AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC
in file.c:waitstream_core().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268578
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Tilghman Lesher [Sun, 6 Jun 2010 05:29:50 +0000 (05:29 +0000)]
Take advantage of variable substitution already in the Makefile to specify the correct location for files in init.d.
(closes issue #16979)
Reported by: jw-asterisk
(issue #15691)
Reported by: itamarjp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268534
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Tilghman Lesher [Sun, 6 Jun 2010 00:37:30 +0000 (00:37 +0000)]
Finally track down and eliminate the "FRACK! warnings from chan_iax2".
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268495
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Tilghman Lesher [Sat, 5 Jun 2010 17:55:28 +0000 (17:55 +0000)]
Fix crash in DTMF detection.
What I did not originally see in my previous commit was that even though the
next digit could be detected before the previous was considered ended, the
detection of the next digit effectively ends the detection of the previous.
Therefore, the length moves in lockstep with the digit, and no separate counter
is needed for the length alone.
(closes issue #17371)
Reported by: alecdavis
(closes issue #17474)
Reported by: kenner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268456
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Tilghman Lesher [Sat, 5 Jun 2010 17:27:12 +0000 (17:27 +0000)]
Verify event is not NULL before attempting to lower its usecount.
(closes issue #17234)
Reported by: mav3rick
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268454
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Kevin P. Fleming [Sat, 5 Jun 2010 05:23:02 +0000 (05:23 +0000)]
Typo fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268417
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Kevin P. Fleming [Sat, 5 Jun 2010 05:12:34 +0000 (05:12 +0000)]
Grammatical error fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268395
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Tilghman Lesher [Sat, 5 Jun 2010 02:51:34 +0000 (02:51 +0000)]
Merged revisions 268320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) | 3 lines
Rest In Peace
http://www.outandaboutnewspaper.com/article/4061
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268321
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David Vossel [Fri, 4 Jun 2010 22:37:13 +0000 (22:37 +0000)]
fixes compile error from uninitialized variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268281
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David Vossel [Fri, 4 Jun 2010 21:55:14 +0000 (21:55 +0000)]
RFC3261 compliant sip unreliable retransmit timing + 'registerattempts' option tweak
Changes.
1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission
timers should initially be set to timer T1. T1 by default is 500ms.
Asterisk was starting the retransmission timers at T1*2 which shouldn't
cause any problems, but is not RFC compliant.
2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction,
if the retransmit timer fires while in the proceeding state that
the request must be retransmitted. Asterisk currently ack's
requests for both INVITE and non-INVITE transactions when a
1XX response is received, this patch changes this for non-INVITE requests.
3. The 'registerattempts' option in sip.conf is supposed to set
how many registry attempts will be made before giving up. When
this option is set to 1, I would expect only one registry attempt
to be made before stopping because of a failure, but instead two are
made. In my opinion this is not expected behavior. This option does
not indicate that these are re-attempts. The logic behind this option
has been changed to only attempt registers the exact number of times
this option is set to. If this option is 0, it still continues to
re-attempt the registration forever.
Review: https://reviewboard.asterisk.org/r/687/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268205
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Tilghman Lesher [Fri, 4 Jun 2010 20:42:27 +0000 (20:42 +0000)]
Merged revisions 268126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04 Jun 2010) | 2 lines
AC_CONFIG_SUBDIRS has a bad side-effect on cross-compiles.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268127
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Tilghman Lesher [Fri, 4 Jun 2010 19:40:00 +0000 (19:40 +0000)]
Merged revisions 268050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04 Jun 2010) | 6 lines
Build menuselect with the build environment's compiler, not the host (target)'s compiler.
(closes issue #17464)
Reported by: pprindeville
Tested by: tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268051
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Tilghman Lesher [Fri, 4 Jun 2010 16:31:25 +0000 (16:31 +0000)]
Merged revisions 267971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010) | 2 lines
As-fixiate the build process
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267972
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Richard Mudgett [Fri, 4 Jun 2010 14:45:03 +0000 (14:45 +0000)]
Incoming overlap dialing no longer works after sig_pri extraction.
The problem would manifest itself if your dialplan matching could accept
more digits to match than were actually dialed. The time out waiting for
overlap digits disconnected the call instead of matching any accumulated
digits to the dialplan.
Accidental conversion of a break out of loop as a break out of switch.
(closes issue #17401)
Reported by: avalentin
Patches:
issue17401_digit_timeout.patch uploaded by rmudgett (license 664)
Tested by: avalentin, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267928
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Tilghman Lesher [Fri, 4 Jun 2010 03:20:47 +0000 (03:20 +0000)]
As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory.
(closes issue #16912)
Reported by: michaelevdokimov
Patches:
asterisk.patch uploaded by michaelevdokimov (license 997)
Tested by: michaelevdokimov
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267877
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Terry Wilson [Fri, 4 Jun 2010 03:11:31 +0000 (03:11 +0000)]
Send an ACK for every final response received for an INVITE
From issue ABE-2247. RFC 3261 compliance for sections 13.2.24 and 17.1.1.2.
Review: https://reviewboard.asterisk.org/r/692/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267863
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Tilghman Lesher [Fri, 4 Jun 2010 02:58:55 +0000 (02:58 +0000)]
As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory.
(closes issue #16912)
Reported by: michaelevdokimov
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267862
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Tilghman Lesher [Fri, 4 Jun 2010 01:36:46 +0000 (01:36 +0000)]
If there's a default, turn it on, even when the option isn't specified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267819
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Tilghman Lesher [Fri, 4 Jun 2010 01:20:17 +0000 (01:20 +0000)]
Merged revisions 267759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) | 7 lines
Make the default install path appear to be /usr on Linux, instead of /usr/local.
Also, reorganize the options, so that they're more alphabetical.
(closes issue #17013)
Reported by: klaus3000
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267775
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Russell Bryant [Thu, 3 Jun 2010 20:41:24 +0000 (20:41 +0000)]
Remove a LOG_WARNING.
This came up when using the sample configs, and just indicates expected behavior.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267714
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Tilghman Lesher [Thu, 3 Jun 2010 19:46:42 +0000 (19:46 +0000)]
Handle OOM errors more gracefully.
(closes issue #17084)
Reported by: falves11
Patches:
issue17084_162_A.diff uploaded by falves11 (license 374)
Tested by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267669
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Leif Madsen [Thu, 3 Jun 2010 18:53:24 +0000 (18:53 +0000)]
Update UPGRADE.txt and CHANGE for CDR functionality changes.
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
(closes issue #17373)
Reported by: wdoekes
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624
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Richard Mudgett [Thu, 3 Jun 2010 18:38:00 +0000 (18:38 +0000)]
Make compile again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267622
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Russell Bryant [Thu, 3 Jun 2010 17:31:41 +0000 (17:31 +0000)]
Don't stop Asterisk if chan_usbradio isn't configured.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267537
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Mark Michelson [Thu, 3 Jun 2010 17:09:11 +0000 (17:09 +0000)]
Remove unnecessary code relating to PLC.
The logic for handling generic PLC is now handled in ast_write in
channel.c instead of in translation code.
Review: https://reviewboard.asterisk.org/r/683/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267492
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Russell Bryant [Thu, 3 Jun 2010 17:05:30 +0000 (17:05 +0000)]
Remove a line that was killing Asterisk on startup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267490
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Russell Bryant [Thu, 3 Jun 2010 14:48:09 +0000 (14:48 +0000)]
Comment out a rule that likes to run implicitly unnecessarily, breaking builds
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267445
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Richard Mudgett [Thu, 3 Jun 2010 00:02:14 +0000 (00:02 +0000)]
Add ETSI Message Waiting Indication (MWI) support.
Add the ability to report waiting messages to ISDN endpoints (phones).
Relevant specification: EN 300 650 and EN 300 745
Review: https://reviewboard.asterisk.org/r/599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399
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Russell Bryant [Wed, 2 Jun 2010 22:46:37 +0000 (22:46 +0000)]
try to fix some random chan_h323 compilation failures
After some debugging, the random chan_h323 build failures appear to be due
to complications introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean commands
down into channels/makefile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267352
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Richard Mudgett [Wed, 2 Jun 2010 22:28:58 +0000 (22:28 +0000)]
Add ETSI Malicious Call ID support.
Add the ability to report malicious callers as an AMI event in the call
event class.
Relevant specification: EN 300 180
Review: https://reviewboard.asterisk.org/r/576/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267350
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Russell Bryant [Wed, 2 Jun 2010 21:44:06 +0000 (21:44 +0000)]
Fix a build error on mac.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267305
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Russell Bryant [Wed, 2 Jun 2010 21:41:54 +0000 (21:41 +0000)]
Ensure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been specified.
When ASTCFLAGS was specified with the make command, Makefile.rules was using
the specified value from the command line and not the one here, making it so this
flag would go missing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267303
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Richard Mudgett [Wed, 2 Jun 2010 21:05:32 +0000 (21:05 +0000)]
Add ETSI Call Waiting support.
Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261
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David Vossel [Wed, 2 Jun 2010 19:33:56 +0000 (19:33 +0000)]
Update CHANGES and aoc help doc to reflect AOC additions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181
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Russell Bryant [Wed, 2 Jun 2010 18:53:38 +0000 (18:53 +0000)]
Add a CLI command that blocks until Asterisk has fully booted.
Review: https://reviewboard.asterisk.org/r/684/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267138
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Mark Michelson [Wed, 2 Jun 2010 18:13:18 +0000 (18:13 +0000)]
Prevent use of uninitialized values.
Two struct sockaddr_ins are created when applying directmedia
host access rules. The addresses of these are passed to the RTP
engine to be filled in. However, the RTP engine inspects the fields
of the structs before actually taking action. This inspection caused
valgrind to be a bit unhappy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267097
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Richard Mudgett [Wed, 2 Jun 2010 18:10:15 +0000 (18:10 +0000)]
Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096
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Russell Bryant [Wed, 2 Jun 2010 17:57:39 +0000 (17:57 +0000)]
Silence a compiler warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267093
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Jeff Peeler [Wed, 2 Jun 2010 17:29:35 +0000 (17:29 +0000)]
Fix infinite loop when loading codec speex
This changes the sample slinear frame data to contain non-zero data so that
translation calculations for speex works when preprocessing and VAD is turned
on. The encoder expects samples to be returned, but when attempted with the
mentioned two options and silent sample frames everything was discarded.
(closes issue #17240)
Reported by: seandarcy
Review: https://reviewboard.asterisk.org/r/682/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267065
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Paul Belanger [Wed, 2 Jun 2010 17:25:05 +0000 (17:25 +0000)]
Merged revisions 267009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun 2010) | 7 lines
Cleanup error/warning messages in AEL2 parser
(closes issue #16684)
Reported by: Silmaril
Patches:
patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267041
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Richard Mudgett [Wed, 2 Jun 2010 17:13:53 +0000 (17:13 +0000)]
Add ETSI Advice Of Charge (AOC) event reporting.
This feature generates AMI events in the new aoc event class from the
events passed up by libpri.
Review: https://reviewboard.asterisk.org/r/537/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267008
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Richard Mudgett [Wed, 2 Jun 2010 16:14:12 +0000 (16:14 +0000)]
Add ETSI Explicit Call Transfer (ECT) support.
Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Review: https://reviewboard.asterisk.org/r/520/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926
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Paul Belanger [Wed, 2 Jun 2010 13:32:22 +0000 (13:32 +0000)]
pthread_join to assure the thread is really gone
(closes issue #15465)
Reported by: fnordian
Patches:
bridging.patch uploaded by fnordian (license 110)
Tested by: lmadsen, fnordian, peterh
Review: https://reviewboard.asterisk.org/r/679/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266877
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Terry Wilson [Tue, 1 Jun 2010 22:14:44 +0000 (22:14 +0000)]
Use the correct ical.h file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266832
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Tilghman Lesher [Tue, 1 Jun 2010 21:28:19 +0000 (21:28 +0000)]
Support setting locale per-mailbox (changes date/time languages for email, pager messages).
(closes issue #14333)
Reported by: klaus3000
Patches:
20090515__issue14333.diff.txt uploaded by tilghman (license 14)
app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266828
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Terry Wilson [Tue, 1 Jun 2010 21:12:49 +0000 (21:12 +0000)]
Set app and appdata fields when a Dial is redirected
(closes issue #17204)
Reported by: one47
Tested by: twilson, one47
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266786
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Tilghman Lesher [Tue, 1 Jun 2010 18:02:24 +0000 (18:02 +0000)]
Don't register functions until the last possible point, so they're not unloaded unnecessarily.
(closes issue #15996)
Reported by: junky
Patches:
sdmi_wait.diff uploaded by junky (license 177)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266735
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Tilghman Lesher [Tue, 1 Jun 2010 16:41:00 +0000 (16:41 +0000)]
Eliminate stale manager events after a set interval, even if AMI clients don't query for them.
Actions (or failures to act) by external clients should not cause memory leaks
in Asterisk, especially when those continued leaks could cause Asterisk to
misbehave later.
(closes issue #17234)
Reported by: mav3rick
Patches:
20100510__issue17234.diff.txt uploaded by tilghman (license 14)
20100517__issue17234__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: mav3rick, davidw
(closes issue #17365)
Reported by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266682
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Tilghman Lesher [Tue, 1 Jun 2010 15:18:59 +0000 (15:18 +0000)]
Merged revisions 266585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines
Prevent CLI prompt from distorting output of lines shorter than the prompt.
Uses the VT100 method of clearing the line from the cursor position to the
end of the line: Esc-0K
(closes issue #17160)
Reported by: coolmig
Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266592
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Tilghman Lesher [Sun, 30 May 2010 20:18:03 +0000 (20:18 +0000)]
Needs to be wrapped in <para>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266522
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Tilghman Lesher [Sun, 30 May 2010 04:44:28 +0000 (04:44 +0000)]
Merged revisions 266437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010) | 2 lines
Reverting patch and reopening issue #16784, as patch breaks color display.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266438
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Terry Wilson [Fri, 28 May 2010 22:54:03 +0000 (22:54 +0000)]
Fix ical library handling (again)
Newer versions of libical (which we require) store the header file in a
libical/ subfolder and include an ical.h file that does a #warning for
deprecation and then #includes <libical/ical.h>. Since we now test for
libical/ical.h, we can change the #includes back to <libical/ical.h> and
remove the test which specifically adds /usr/include/libical as an include
directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266386
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Tilghman Lesher [Fri, 28 May 2010 22:50:06 +0000 (22:50 +0000)]
Setup environment variables for the benefit of child processes and disallow changing them.
(closes issue #14899)
Reported by: jmls
Patches:
20090916__issue14899.diff.txt uploaded by tilghman (license 14)
Tested by: jmls
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266385
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Tilghman Lesher [Fri, 28 May 2010 20:53:04 +0000 (20:53 +0000)]
Only report swap on platforms which can examine those statistics
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266337
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David Vossel [Fri, 28 May 2010 17:55:38 +0000 (17:55 +0000)]
fixes crash when creation of UDPTL fails
(closes issue #17264)
Reported by: falves11
Patches:
issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671)
issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671)
Tested by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266292
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Terry Wilson [Fri, 28 May 2010 17:34:00 +0000 (17:34 +0000)]
More build fixes for ical/neon and res_calendar_ews
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266289
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Jeff Peeler [Thu, 27 May 2010 20:08:49 +0000 (20:08 +0000)]
fix compile error
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266240
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Tilghman Lesher [Thu, 27 May 2010 19:25:16 +0000 (19:25 +0000)]
Cache query results for one second.
Queries from the PBX core come in 3's. Caching avoids the additional
performance penalty from those two additional queries hitting the database.
(closes issue #16521)
Reported by: tilghman
Patches:
20091229__issue16521.diff.txt uploaded by tilghman (license 14)
Tested by: Hubguru, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266238
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Tilghman Lesher [Wed, 26 May 2010 21:17:46 +0000 (21:17 +0000)]
Merged revisions 266142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines
Use sigaction for signals which should persist past the initial trigger, not signal.
If you call signal() in a Solaris signal handler, instead of just resetting
the signal handler, it causes the signal to refire, because the signal is not
marked as handled prior to the signal handler being called. This effectively
causes Solaris to immediately exceed the threadstack in recursive signal
handlers and crash.
(closes issue #17000)
Reported by: rmcgilvr
Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266146
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Mark Michelson [Wed, 26 May 2010 20:17:54 +0000 (20:17 +0000)]
Remove redundant ast_conntected_line_free call.
This wouldn't cause any problems, but it's certainly not needed either.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266098
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Mark Michelson [Wed, 26 May 2010 20:15:47 +0000 (20:15 +0000)]
Remove unrelated MOH change from previous commit.
Thanks Kevin!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266094
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Mark Michelson [Wed, 26 May 2010 20:04:51 +0000 (20:04 +0000)]
Fix misspelling of macro args.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266092
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David Vossel [Wed, 26 May 2010 19:46:49 +0000 (19:46 +0000)]
do all sip registry parsing before transmit_register
This patch breaks up every part of the sip registry string during
config parsing and removes all parsing from transmit_register().
Thanks to Nick_Lewis for contributing this patch!
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657)
chan_sip.c.patch uploaded by Nick Lewis (license 657)
chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657)
nicklewispatch.diff uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
Review: https://reviewboard.asterisk.org/r/628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090
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David Vossel [Wed, 26 May 2010 18:32:51 +0000 (18:32 +0000)]
fixes failed SIP Directed pickup resulting in dead channel
(closes issue #17339)
Reported by: one47
Patches:
sip_magic_pickup2 uploaded by one47 (license 23)
Tested by: one47, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266006
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Mark Michelson [Wed, 26 May 2010 18:21:53 +0000 (18:21 +0000)]
Blocked revisions 266004 via svnmerge
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r266004 | mmichelson | 2010-05-26 13:21:10 -0500 (Wed, 26 May 2010) | 12 lines
Make AgentComplete message more consistent.
At times, the "Member" field was not specified during the event.
It's there now.
(closes issue #15638)
Reported by: elbriga
Patches:
patchAppQueueAgentComplete.diff uploaded by elbriga (license 482)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266005
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Tilghman Lesher [Wed, 26 May 2010 16:23:28 +0000 (16:23 +0000)]
Merged revisions 265910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines
Not finding rows in the DB does not rise to the level of a warning.
(closes issue #17062)
Reported by: drookie
Patches:
20100525__issue17062.diff.txt uploaded by tilghman (license 14)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265923
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Tilghman Lesher [Wed, 26 May 2010 16:14:48 +0000 (16:14 +0000)]
Construct socket name, according to the Postgres docs, and document as such.
(closes issue #17392)
Reported by: dps
Patches:
20100525__issue17392.diff.txt uploaded by tilghman (license 14)
Tested by: dps
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265894
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Mark Michelson [Wed, 26 May 2010 14:45:47 +0000 (14:45 +0000)]
.......
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265844
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Mark Michelson [Wed, 26 May 2010 14:41:55 +0000 (14:41 +0000)]
Re-enable "always" option for videosupport option in sip.conf.
(closes issue #17016)
Reported by: twilson
Patches:
17016.patch uploaded by mmichelson (license 60)
Tested by: devmod
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265842
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Terry Wilson [Wed, 26 May 2010 05:33:11 +0000 (05:33 +0000)]
Ensure that libneon > 0.29.0 is installed for res_calendar_ews
This uses a modified version of pabelanger's patch that checks for NTLM support
instead, which was added in 0.29.0 which is what is required for
res_calendar_ews.
(closes issue #17391)
Reported by: loloski
Patches:
issue17391.patch.v2 uploaded by pabelanger (license 224)
Tested by: twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793
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Tilghman Lesher [Wed, 26 May 2010 00:29:40 +0000 (00:29 +0000)]
Use configure to determine the prefixes and include directories properly.
This ensures cross-platform compatibility, even among Linux distributions,
which don't always put headers in the same place.
(closes issue #17391)
Reported by: loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265747
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Mark Michelson [Tue, 25 May 2010 20:59:04 +0000 (20:59 +0000)]
Properly use peer's outboundproxy for outbound REGISTERs.
The logic used in transmit_register to get the outboundproxy for a peer
was flawed since this value would be overridden shortly afterwards when
create_addr was called.
In addition, this also fixes some logic used when parsing users.conf so
that the peer name is placed in the internally-generated register string
so that an outboundproxy set in the Asterisk GUI will be used for outbound
REGISTERs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265698
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David Vossel [Tue, 25 May 2010 17:12:28 +0000 (17:12 +0000)]
Blocked revisions 265613 via svnmerge
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r265613 | dvossel | 2010-05-25 12:11:40 -0500 (Tue, 25 May 2010) | 8 lines
fixes build issue with zaptel
(closes issue #17394)
Reported by: aragon
Patches:
half_buffer_fix.diff uploaded by dvossel (license 671)
Tested by: aragon
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265614
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Matthew Nicholson [Tue, 25 May 2010 17:00:11 +0000 (17:00 +0000)]
Merged revisions 265610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines
Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670.
(closes issue #17334)
Reported by: jvandal
Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
Tested by: aragon, jvandal
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265611
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Richard Mudgett [Tue, 25 May 2010 16:23:51 +0000 (16:23 +0000)]
Memory leak in connected line data when SIP blond transfer done.
The handling of the control subclass AST_CONTROL_READ_ACTION frame leaked
connected line string memory in __ast_read().
Also in __ast_read() the frame type switch should not have had a case for
AST_CONTROL_READ_ACTION. AST_CONTROL_READ_ACTION is not a frame type.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265608
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Tzafrir Cohen [Tue, 25 May 2010 08:31:51 +0000 (08:31 +0000)]
Typos: 'succesful' (lintian)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265525
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Terry Wilson [Mon, 24 May 2010 22:21:58 +0000 (22:21 +0000)]
Merge the rest of the FullyBooted patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467
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