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git.ipfire.org Git - thirdparty/asterisk.git/log
Brett Bryant [Fri, 3 Sep 2010 18:21:47 +0000 (18:21 +0000)]
Merged revisions 284967 via svnmerge from
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r284967 | bbryant | 2010-09-03 14:19:53 -0400 (Fri, 03 Sep 2010) | 15 lines
Merged revisions 284958 via svnmerge from
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r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) | 8 lines
This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response.
(closes issue #17935)
Reported by: alexkuklin
Patches:
iaxshowreg uploaded by alexkuklin (license 1115)
Tested by: alexkuklin
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David Vossel [Fri, 3 Sep 2010 18:04:10 +0000 (18:04 +0000)]
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r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) | 2 lines
During OPTIONS authentication, the authpeer does not need to be returned for any reason.
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David Vossel [Fri, 3 Sep 2010 17:30:04 +0000 (17:30 +0000)]
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r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines
authenticate OPTIONS requests just like we would an INVITE
OPTIONS requests should be treated the same as an INVITE
This includes authentication. This patch adds the ability for
incoming out of dialog OPTION requests to be authenticated
before providing a response indicating whether an extension
is available or not. The authentication routine works the
exact same way as it does for incoming INVITEs. This means
that if a peer has 'insecure=invite' in their peer definition,
the same will be true for the processing of the OPTIONS request.
Review: https://reviewboard.asterisk.org/r/881/
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Terry Wilson [Fri, 3 Sep 2010 16:42:53 +0000 (16:42 +0000)]
Merged revisions 284921 via svnmerge from
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r284921 | twilson | 2010-09-03 11:28:18 -0500 (Fri, 03 Sep 2010) | 19 lines
Merged revisions 284897 via svnmerge from
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r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines
Merged revisions 284881 via svnmerge from
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r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
code treated missing files as though they existed.
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Jan Kalab [Fri, 3 Sep 2010 13:09:23 +0000 (13:09 +0000)]
Merge of strdupa() fix for calendars categories priorities
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284853
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Jan Kalab [Fri, 3 Sep 2010 12:58:52 +0000 (12:58 +0000)]
Support for calendar events priorities and categories (with ISO C90 fix)
See RFC 5545 ch. 3.8.1.2 and 9.
(closes issue #17837)
Review: https://reviewboard.asterisk.org/r/880/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284851
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Richard Mudgett [Thu, 2 Sep 2010 21:08:41 +0000 (21:08 +0000)]
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r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) | 8 lines
Made output libpri event names if pri debugging is enabled when sig_pri processes them.
* Simplified CLI "pri debug xx span xx" command code and removed redundant
debugging enabled messages.
* Made CLI "pri debug xx span xx" command only close the debugging log
file if it was opened.
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r284780 | rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines
Simplified pri_dchannel() poll timeout duration code.
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David Vossel [Thu, 2 Sep 2010 16:57:43 +0000 (16:57 +0000)]
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r284705 | dvossel | 2010-09-02 11:56:43 -0500 (Thu, 02 Sep 2010) | 20 lines
Merged revisions 284704 via svnmerge from
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r284704 | dvossel | 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) | 13 lines
Merged revisions 284703 via svnmerge from
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r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines
Removed relatedpeer code from sip_autodestruct
Handling of the relatedpeer structure associated with a
sip_pvt should be done during the final sip_destruction
function, not in sip_autodestruct.
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Jason Parker [Thu, 2 Sep 2010 16:44:06 +0000 (16:44 +0000)]
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r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep 2010) | 8 lines
Add slin16 support for format_wav (new wav16 file extension)
(closes issue #15029)
Reported by: andrew
Patches:
wav16.patch uploaded by andrew (license 240)
Tested by: qwell, andrew
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Tilghman Lesher [Thu, 2 Sep 2010 16:36:55 +0000 (16:36 +0000)]
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r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010) | 2 lines
Fixing build
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Richard Mudgett [Thu, 2 Sep 2010 16:35:39 +0000 (16:35 +0000)]
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r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) | 5 lines
Added documentation for CONNECTEDLINE and REDIRECTING functions.
(closes issue #17808)
Reported by: jtodd
Review: https://reviewboard.asterisk.org/r/875/
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Tilghman Lesher [Thu, 2 Sep 2010 16:12:34 +0000 (16:12 +0000)]
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r284666 | tilghman | 2010-09-02 11:11:15 -0500 (Thu, 02 Sep 2010) | 9 lines
Merged revisions 284665 via svnmerge from
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r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) | 2 lines
Fixing build.
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Tilghman Lesher [Thu, 2 Sep 2010 05:31:47 +0000 (05:31 +0000)]
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r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines
Merged revisions 284631 via svnmerge from
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r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
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Tilghman Lesher [Thu, 2 Sep 2010 05:27:53 +0000 (05:27 +0000)]
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r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707)
Reported by: ira
Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/876/
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Tilghman Lesher [Thu, 2 Sep 2010 05:02:54 +0000 (05:02 +0000)]
Merged revisions 284597 via svnmerge from
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r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
Merged revisions 284593,284595 via svnmerge from
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r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
Merged revisions 284478 via svnmerge from
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
Failed to rerun bootstrap.sh after last commit
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David Vossel [Wed, 1 Sep 2010 21:48:32 +0000 (21:48 +0000)]
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r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines
During request to dialog matching, verify init_ruri is present before comparing.
During request to dialog matching, we attempt a best effort routine for fork
detection which requires several elements to be in place. The dialog's
initial request uri is one of those elements. Since it is best effort,
if the init_ruri is not present for some reason we can not proceed with that
routine.
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Terry Wilson [Wed, 1 Sep 2010 18:52:27 +0000 (18:52 +0000)]
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r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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Tilghman Lesher [Wed, 1 Sep 2010 18:19:12 +0000 (18:19 +0000)]
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r284473 | tilghman | 2010-09-01 13:16:37 -0500 (Wed, 01 Sep 2010) | 12 lines
Merged revisions 284472 via svnmerge from
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r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) | 5 lines
Don't warn on floats and timestamps
(closes issue #17082)
Reported by: coolmig
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Tilghman Lesher [Tue, 31 Aug 2010 20:47:28 +0000 (20:47 +0000)]
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r284415 | tilghman | 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) | 21 lines
Merged revisions 284399 via svnmerge from
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r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines
Merged revisions 284393 via svnmerge from
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r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines
Don't send a devstate change on poke_noanswer if the state did not change.
(closes issue #17741)
Reported by: schmidts
Patches:
chan_sip.c.patch uploaded by schmidts (license 1077)
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Leif Madsen [Tue, 31 Aug 2010 19:01:12 +0000 (19:01 +0000)]
Merged revisions 284318 via svnmerge from
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r284318 | lmadsen | 2010-08-31 14:00:15 -0500 (Tue, 31 Aug 2010) | 22 lines
Merged revisions 284317 via svnmerge from
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r284317 | lmadsen | 2010-08-31 13:59:31 -0500 (Tue, 31 Aug 2010) | 15 lines
Merged revisions 284316 via svnmerge from
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r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines
Update say.conf.sample to match the rules in say.c
(closes issue #17835)
Reported by: RoadKill
Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill
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Leif Madsen [Tue, 31 Aug 2010 18:53:51 +0000 (18:53 +0000)]
Add trustrpid and sendrpid global values to 'sip show settings'
(closes issue #17860)
Reported by: jtodd
Patches:
__20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, russell
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Tilghman Lesher [Mon, 30 Aug 2010 22:30:10 +0000 (22:30 +0000)]
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r284281 | tilghman | 2010-08-30 17:28:47 -0500 (Mon, 30 Aug 2010) | 18 lines
Merged revisions 284280 via svnmerge from
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r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines
Fix 3 coding errors:
1) After we close FD, we should not be trying to write to it.
2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child.
3) Use endian, not processor, detection to ensure bytes are written in the correct order.
(closes issue #15706)
Reported by: modelnine
Patches:
asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865)
Tested by: gmartinez
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Olle Johansson [Mon, 30 Aug 2010 09:32:17 +0000 (09:32 +0000)]
Small doxygen fix and doc addition
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Olle Johansson [Mon, 30 Aug 2010 09:29:03 +0000 (09:29 +0000)]
Clean upp doxygen documentation
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Olle Johansson [Mon, 30 Aug 2010 08:25:50 +0000 (08:25 +0000)]
Doxygen formatting
You can't write "same as above" in hypertext documentation. Above doesn't make sense in
hyperspace.
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Olle Johansson [Mon, 30 Aug 2010 08:03:42 +0000 (08:03 +0000)]
Add doxygen documentation
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Tilghman Lesher [Sun, 29 Aug 2010 07:06:08 +0000 (07:06 +0000)]
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r284158 | tilghman | 2010-08-29 02:05:27 -0500 (Sun, 29 Aug 2010) | 2 lines
Missed adding this file
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Tilghman Lesher [Sun, 29 Aug 2010 05:18:18 +0000 (05:18 +0000)]
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r284127 | tilghman | 2010-08-29 00:17:37 -0500 (Sun, 29 Aug 2010) | 2 lines
Also ignore the checksums
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Tilghman Lesher [Sun, 29 Aug 2010 02:52:25 +0000 (02:52 +0000)]
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r284096 | tilghman | 2010-08-28 21:51:14 -0500 (Sat, 28 Aug 2010) | 3 lines
Rename CEL adaptive driver to plain driver, since there isn't another ODBC driver
(and the other CEL drivers have adaptive capabilities, anyway).
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Russell Bryant [Sat, 28 Aug 2010 21:30:25 +0000 (21:30 +0000)]
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r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
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David Vossel [Fri, 27 Aug 2010 22:39:48 +0000 (22:39 +0000)]
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r284032 | dvossel | 2010-08-27 17:37:11 -0500 (Fri, 27 Aug 2010) | 21 lines
Merged revisions 284002 via svnmerge from
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r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines
Merged revisions 283960 via svnmerge from
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r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines
Parse all "Accept" headers for SIP SUBSCRIBE requests.
(closes issue #17758)
Reported by: ibc
Patches:
multiple_accept_headers_1.4.diff uploaded by dvossel (license 671)
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Russell Bryant [Fri, 27 Aug 2010 21:50:15 +0000 (21:50 +0000)]
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r283951 | russell | 2010-08-27 16:33:55 -0500 (Fri, 27 Aug 2010) | 2 lines
Print exten@context:priority in verbose messages from pbx_realtime.
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Jason Parker [Fri, 27 Aug 2010 20:32:21 +0000 (20:32 +0000)]
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r283882 | qwell | 2010-08-27 15:31:55 -0500 (Fri, 27 Aug 2010) | 22 lines
Merged revisions 283881 via svnmerge from
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r283881 | qwell | 2010-08-27 15:30:27 -0500 (Fri, 27 Aug 2010) | 15 lines
Merged revisions 283880 via svnmerge from
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r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines
Fix issue with decoding ^-escaped characters in realtime.
(closes issue #17790)
Reported by: denzs
Patches:
17790-chunky.diff uploaded by qwell (license 4)
Tested by: qwell, denzs
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Olle Johansson [Fri, 27 Aug 2010 14:01:21 +0000 (14:01 +0000)]
Doxygen formatting changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283803
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Tilghman Lesher [Thu, 26 Aug 2010 23:51:55 +0000 (23:51 +0000)]
Merged revisions 283770 via svnmerge from
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r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26 Aug 2010) | 8 lines
Convert MOH to use generic timers.
(closes issue #17726)
Reported by: lmadsen
Patches:
20100825__issue17726__2.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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David Vossel [Thu, 26 Aug 2010 15:28:07 +0000 (15:28 +0000)]
Merged revisions 283692 via svnmerge from
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r283692 | dvossel | 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) | 32 lines
Merged revisions 283691 via svnmerge from
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r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines
Merged revisions 283690 via svnmerge from
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r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines
Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response.
If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response
to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc
compliant and results in confusion at the other endpoint. sip_pretend_ack will ack
and remove all the packets in the retransmit queue. This means that the INVITE will
stop retransmitting, and that any response to that INVITE that comes after the pretend_ack
occurs will be ignored.
Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal
hangup, we should let the protocol stack process the INVITE transaction and terminate
the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag
is used, once the dialog proceeds to an escapable state the transaction will either be
canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do
this any other way is incorrect. If the endpoint is not responding to the INVITE request,
the INVITE must continue to be retransmitted until it times out which will result in the
dialog being destroyed.
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Russell Bryant [Thu, 26 Aug 2010 13:28:05 +0000 (13:28 +0000)]
Merged revisions 283659 via svnmerge from
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r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010) | 2 lines
Slight improvement to a debug message.
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Russell Bryant [Thu, 26 Aug 2010 12:49:09 +0000 (12:49 +0000)]
Merged revisions 283629 via svnmerge from
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r283629 | russell | 2010-08-26 07:48:45 -0500 (Thu, 26 Aug 2010) | 2 lines
Remove public keys that are no longer useful.
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Russell Bryant [Thu, 26 Aug 2010 12:26:46 +0000 (12:26 +0000)]
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r283627 | russell | 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010) | 2 lines
Move httptimeout out from in between port and bindaddr.
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David Vossel [Wed, 25 Aug 2010 22:59:15 +0000 (22:59 +0000)]
Merged revisions 283595 via svnmerge from
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r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines
Merged revisions 283594 via svnmerge from
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r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines
Add to and from tags to NOTIFY dialog-info xml body so pickup can occur.
When pedantic mode is used, the dialog-info xml generated during a
ringing event must contain the to and from tag values. Otherwise if
a pickup occurs using INVITE with replaces, Astrisk will not be able
to locate the subscription.
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Tilghman Lesher [Wed, 25 Aug 2010 16:14:11 +0000 (16:14 +0000)]
Merged revisions 283561 via svnmerge from
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r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) | 5 lines
Initialize connect timeout on each time through the loop.
(closes issue #17911)
Reported by: wurstsalat
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David Vossel [Wed, 25 Aug 2010 15:56:05 +0000 (15:56 +0000)]
Merged revisions 283559 via svnmerge from
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r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines
Merged revisions 283558 via svnmerge from
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r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines
Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
Asterisk now dynamically builds the "Supported" header depending
on what is enabled/disabled in sip.conf. Session timers used
to always be advertised as being supported even when they were disabled
in the configuration. This caused problems with some end points.
(issue #17005)
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Russell Bryant [Wed, 25 Aug 2010 14:55:47 +0000 (14:55 +0000)]
Merged revisions 283527 via svnmerge from
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r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines
Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
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Damien Wedhorn [Tue, 24 Aug 2010 20:42:30 +0000 (20:42 +0000)]
Ignore redial hard button when no previous number.
(closes issue #17887)
Reported by: salecha
Patches:
skinny.redial.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
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David Vossel [Tue, 24 Aug 2010 20:36:35 +0000 (20:36 +0000)]
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r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines
Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
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Leif Madsen [Tue, 24 Aug 2010 18:58:46 +0000 (18:58 +0000)]
Merged revisions 283457 via svnmerge from
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r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines
Fix issue where TOS is no longer set on RTP packets.
Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk.
(closes issue #17890)
Reported by: elguero
Patches:
qos_18.diff uploaded by elguero (license 37)
Review: https://reviewboard.asterisk.org/r/868
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David Vossel [Tue, 24 Aug 2010 18:45:57 +0000 (18:45 +0000)]
This fix downgrades the ERROR message indicating no res_stun_monitor.conf to a WARNING message.
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David Vossel [Tue, 24 Aug 2010 16:12:36 +0000 (16:12 +0000)]
Merged revisions 283382 via svnmerge from
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r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines
Merged revisions 283381 via svnmerge from
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r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines
Merged revisions 283380 via svnmerge from
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r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines
This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
When the pending bye flag is used, it is possible that the dialog will terminate
and leave the sip_pvt->owner channel up. This is because we never hangup the
ast_channel after sending the SIP_BYE request. When we receive the response for
the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the
next do_monitor loop, but this is not the case. The dialog will only be destroyed
once the owner is hungup even with the need_destroy flag set. This patch sets the
softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the
pending bye flag.
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Russell Bryant [Tue, 24 Aug 2010 12:51:46 +0000 (12:51 +0000)]
Merged revisions 283350 via svnmerge from
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r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) | 2 lines
Don't attempt to release a NULL ODBC handle.
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Tilghman Lesher [Mon, 23 Aug 2010 21:35:09 +0000 (21:35 +0000)]
Merged revisions 283319 via svnmerge from
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r283319 | tilghman | 2010-08-23 16:33:47 -0500 (Mon, 23 Aug 2010) | 9 lines
Merged revisions 283318 via svnmerge from
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r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010) | 2 lines
CDR drivers depend upon res_odbc, not directly on the ODBC libraries
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Damien Wedhorn [Mon, 23 Aug 2010 20:50:55 +0000 (20:50 +0000)]
Hack to allow easy debugging of skinny in trunk.
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Damien Wedhorn [Mon, 23 Aug 2010 20:39:05 +0000 (20:39 +0000)]
Add additional AST_CONTROL_ states to control2str.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283288
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Damien Wedhorn [Mon, 23 Aug 2010 20:23:51 +0000 (20:23 +0000)]
Fixes display issues on 7910 and older phones.
Also correct the callinfo provided in skinny_answer.
(closes issue #17876)
Reported by: salecha
Patches:
skinny_cnd3.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283287
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Russell Bryant [Mon, 23 Aug 2010 13:35:55 +0000 (13:35 +0000)]
Merged revisions 283241 via svnmerge from
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r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23 Aug 2010) | 2 lines
Add sample configuration for cel_radius.
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Russell Bryant [Mon, 23 Aug 2010 13:23:37 +0000 (13:23 +0000)]
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r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by
the crash that I hit because there was an assumption in the code that the
values in each matched up. However, this is a quick fix to get them to
match up so it will work.
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Russell Bryant [Mon, 23 Aug 2010 13:09:47 +0000 (13:09 +0000)]
Merged revisions 283209 via svnmerge from
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r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) | 2 lines
Don't blow up on an invalid AMA flag.
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Russell Bryant [Mon, 23 Aug 2010 12:33:24 +0000 (12:33 +0000)]
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r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010) | 2 lines
Tack on ${eventextra} to the sample cel_custom.conf.
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Russell Bryant [Mon, 23 Aug 2010 12:13:16 +0000 (12:13 +0000)]
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r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010) | 2 lines
Cut down on excessive quotation.
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Tilghman Lesher [Mon, 23 Aug 2010 12:09:17 +0000 (12:09 +0000)]
Merged revisions 283175 via svnmerge from
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r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 Aug 2010) | 2 lines
Don't fail to start if the config file is missing.
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Russell Bryant [Mon, 23 Aug 2010 11:59:51 +0000 (11:59 +0000)]
Merged revisions 283173 via svnmerge from
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r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) | 5 lines
Expand cel_custom.conf.sample.
Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list
the special CEL variables that are available for use in the mapping.
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Richard Mudgett [Fri, 20 Aug 2010 15:39:25 +0000 (15:39 +0000)]
Merged revisions 283050 via svnmerge from
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r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines
Merged revisions 283049 via svnmerge from
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r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines
Merged revisions 283048 via svnmerge from
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r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines
Q931 - Sending PROGRESS after sending ALERTING is a protocol error
The PRI layer in chan_dadhi will check if a PROGRESS message has already
been sent, and not allow sending another (although that is technically
allowed by the Q931 spec), however it does not protect against sending an
ALERTING and then sending a PROGRESS message, which is a violation of the
specification.
Most switches don't seem to care too deeply about this, but some do, and
will disconnect the call when receiving this invalid sequence.
Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure
A.5/Q.931 -- Overview protocol control (network side) point-point
(sheet 3 of 8)"
(closes issue #17874)
Reported by: nic_bellamy
Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
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Russell Bryant [Fri, 20 Aug 2010 12:45:30 +0000 (12:45 +0000)]
Merged revisions 283013 via svnmerge from
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r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010) | 2 lines
Fix a typo in a column name.
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Russell Bryant [Fri, 20 Aug 2010 11:54:22 +0000 (11:54 +0000)]
Merged revisions 282979 via svnmerge from
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r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010) | 2 lines
Add an argument missing from the CELGenUserEvent documentation.
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Russell Bryant [Fri, 20 Aug 2010 11:54:00 +0000 (11:54 +0000)]
Merged revisions 282638 via svnmerge from
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r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) | 4 lines
Split _all_ arguments before parsing them.
This fixes multicast RTP paging using linksys mode.
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David Vossel [Thu, 19 Aug 2010 21:08:39 +0000 (21:08 +0000)]
Merged revisions 282895 via svnmerge from
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r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines
Merged revisions 282894 via svnmerge from
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r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines
Merged revisions 282893 via svnmerge from
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r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines
tos_sip option was not being set correctly
When tos_sip is used, the tos of the sip socket is only set
correctly if the socket binding changes on a reload. If the binding
stays the same but the TOS changes, the new tos value would not take
into effect. This patch fixes that.
(closes issue #17712)
Reported by: nickb
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David Vossel [Thu, 19 Aug 2010 20:35:42 +0000 (20:35 +0000)]
Merged revisions 282891 via svnmerge from
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r282891 | dvossel | 2010-08-19 15:34:41 -0500 (Thu, 19 Aug 2010) | 11 lines
Merged revisions 282890 via svnmerge from
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r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines
fixes sip peer memory leaks in the peer_by_ip table
(issue #17798)
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Matthew Nicholson [Thu, 19 Aug 2010 20:02:52 +0000 (20:02 +0000)]
Merged revisions 282860 via svnmerge from
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r282860 | mnicholson | 2010-08-19 15:01:11 -0500 (Thu, 19 Aug 2010) | 30 lines
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r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines
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r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines
Regression with T.38 negotiation
Prior to 1.4.26.3 T.38 negotiation worked properly, in the case
of the reporter.
(issue #16852)
Reported by: cfc
(closes issue #16705)
Reported by: mpiazzatnetbug
Patches:
issue16705_2.diff uploaded by ebroad (license 878)
Tested by: vrban, ebroad, c0rnoTa, samdell3
Review: https://reviewboard.asterisk.org/r/754/
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Tilghman Lesher [Thu, 19 Aug 2010 14:46:08 +0000 (14:46 +0000)]
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r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) | 2 lines
Only output debugging if the debug level is on.
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Russell Bryant [Thu, 19 Aug 2010 12:13:41 +0000 (12:13 +0000)]
Add a todo item for CEL.
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Terry Wilson [Thu, 19 Aug 2010 02:20:42 +0000 (02:20 +0000)]
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r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines
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r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines
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r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
Add some documentation about codec negotiation to sip.conf
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Damien Wedhorn [Wed, 18 Aug 2010 21:34:27 +0000 (21:34 +0000)]
Cleanup: consolidate offhook (new call).
Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one.
(closes issue #17812)
Reported by: wedhorn
Patches:
cleanup.stateoffhook.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
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Richard Mudgett [Wed, 18 Aug 2010 15:35:23 +0000 (15:35 +0000)]
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r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) | 1 line
Use the correct operator when calculating the PRI span devstate.
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r282672 | rmudgett | 2010-08-18 10:28:27 -0500 (Wed, 18 Aug 2010) | 1 line
Use the correct type for aoce_delayhangup bit field.
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Matthew Nicholson [Wed, 18 Aug 2010 13:11:38 +0000 (13:11 +0000)]
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r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines
Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests.
This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received.
(issue #17486)
Reported by: davidw
Tested by: mnicholson
(issue #12713)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/860/
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Tilghman Lesher [Wed, 18 Aug 2010 07:50:07 +0000 (07:50 +0000)]
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r282608 | tilghman | 2010-08-18 02:49:04 -0500 (Wed, 18 Aug 2010) | 16 lines
Merged revisions 282607 via svnmerge from
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r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines
Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes.
(closes issue #16770)
Reported by: jamicque
Patches:
20100413__issue16770.diff.txt uploaded by tilghman (license 14)
20100811__issue16770.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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David Vossel [Tue, 17 Aug 2010 21:37:46 +0000 (21:37 +0000)]
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r282577 | dvossel | 2010-08-17 16:36:57 -0500 (Tue, 17 Aug 2010) | 16 lines
Merged revisions 282576 via svnmerge from
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r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines
fixes no default transport for temp peer creation in chan_sip
(closes issue #17829)
Reported by: falves11
Patches:
issue_17829.rev1.txt uploaded by russell (license 2)
issue_17829.diff uploaded by dvossel (license 671)
Tested by: falves11
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David Vossel [Tue, 17 Aug 2010 20:09:30 +0000 (20:09 +0000)]
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r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 Aug 2010) | 6 lines
ACCEPT message should respond with the new FORMAT2 ie
(closes issue #17804)
Reported by: tpanton
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David Vossel [Tue, 17 Aug 2010 19:34:52 +0000 (19:34 +0000)]
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r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) | 4 lines
fixes truncated uint64_t value in put_unaligned_uint64_t() function
(issue #17804)
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Terry Wilson [Mon, 16 Aug 2010 20:40:55 +0000 (20:40 +0000)]
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r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines
Merged revisions 282467 via svnmerge from
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r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines
Merged revisions 282430 via svnmerge from
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r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines
Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is potentially
changing, so send a SRCCHANGE so that RTP-based media streams can get
a new SSRC generated to reflect the change. Original patch by addix
(along with lots of testing--thanks!).
(closes issue #17007)
Reported by: addix
Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006)
srcchange.diff uploaded by twilson (license 396)
Tested by: addix, twilson
Review: https://reviewboard.asterisk.org/r/862/
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Leif Madsen [Mon, 16 Aug 2010 18:02:29 +0000 (18:02 +0000)]
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r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
Merged revisions 282469 via svnmerge from
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r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
Add information about creating sounds files using
the sounds tools publically available so that others can create their
own sounds prompts using the same tools we use to generate sounds releases.
This allows people creating their own prompts to sound consistent with
the prompts available from the open source project.
SWP-595
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Tzafrir Cohen [Sun, 15 Aug 2010 13:08:45 +0000 (13:08 +0000)]
Support for GNU/kFreeBSD
kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See
http://glibc-bsd.alioth.debian.org/porting/PORTING
This patch gets Asterisk close to building on Debian kFreeBSD i386,
mainly by adding an extra test for __GLIBC__ in one or two (or more)
places.
OSARCH is set to 'kfreebsd-gnu'
DAHDI support (and support for chan_vpb) was not tested.
Review: https://reviewboard.asterisk.org/r/858/
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Tilghman Lesher [Sat, 14 Aug 2010 04:58:34 +0000 (04:58 +0000)]
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r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines
Fix our FRACKing issue with chan_iax2 a different way.
Review: https://reviewboard.asterisk.org/r/861/
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Richard Mudgett [Fri, 13 Aug 2010 23:57:21 +0000 (23:57 +0000)]
Merged revisions 282334 via svnmerge from
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r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines
PRI CCSS may use a stale dial string for the recall dial string.
If an outgoing call negotiates a different B channel than initially
requested, the saved original dial string was not transferred to the new B
channel. CCSS uses that dial string to generate the recall dial string.
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David Vossel [Fri, 13 Aug 2010 22:27:20 +0000 (22:27 +0000)]
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r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
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David Vossel [Fri, 13 Aug 2010 20:12:22 +0000 (20:12 +0000)]
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r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines
res_stun_monitor and corresponding options CHANGES documentation
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David Vossel [Fri, 13 Aug 2010 20:05:44 +0000 (20:05 +0000)]
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r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854
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David Vossel [Fri, 13 Aug 2010 18:58:49 +0000 (18:58 +0000)]
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r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
Merged revisions 282235 via svnmerge from
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r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
a subscriber that a device is ringing. This option should only be enabled
when the new 'notifycid' option is set... but this was not the case. Instead
the call-id value was included for every RINGING Notify message, which
caused a regression for people who used other methods for call pickup.
(closes issue #17633)
Reported by: urosh
Patches:
chan_sip.txt uploaded by urosh (license )
blf_cid_issue.diff uploaded by dvossel (license 671)
Tested by: dvossel, urosh, okrief, alecdavis
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Terry Wilson [Fri, 13 Aug 2010 16:08:05 +0000 (16:08 +0000)]
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r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines
Detect when libsrtp cannot be linked in a shared library
The libsrtp build system currently does not produce a shared library
or a static library compiled with -fPIC, so on 64-bit systems it is
possible that we will get a compile error if libsrtp is installed and
res_srtp is selected in menuselect.
This patch attempts to detect this situation and provide the user with
instructions to work around the problem.
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r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
Whitespace fix :-/
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Jason Parker [Thu, 12 Aug 2010 22:52:33 +0000 (22:52 +0000)]
Merged revisions 282131 via svnmerge from
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r282131 | qwell | 2010-08-12 17:51:44 -0500 (Thu, 12 Aug 2010) | 16 lines
Merged revisions 282130 via svnmerge from
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r282130 | qwell | 2010-08-12 17:50:54 -0500 (Thu, 12 Aug 2010) | 9 lines
Merged revisions 282129 via svnmerge from
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r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line
Register CLI commands before parsing config, in case there is a config error.
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Richard Mudgett [Thu, 12 Aug 2010 22:10:49 +0000 (22:10 +0000)]
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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Russell Bryant [Thu, 12 Aug 2010 20:44:39 +0000 (20:44 +0000)]
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r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines
Add a "core reload" CLI command.
Review: https://reviewboard.asterisk.org/r/859/
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David Vossel [Thu, 12 Aug 2010 20:17:17 +0000 (20:17 +0000)]
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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Russell Bryant [Thu, 12 Aug 2010 18:04:19 +0000 (18:04 +0000)]
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r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) | 2 lines
Put back pointer value output for ast_debug(), such that it is only removed for verbose output.
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Russell Bryant [Thu, 12 Aug 2010 16:48:54 +0000 (16:48 +0000)]
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r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines
Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to users in the
verbose messages about adding extensions and contexts.
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Jeff Peeler [Thu, 12 Aug 2010 03:08:45 +0000 (03:08 +0000)]
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r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines
Merged revisions 281912 via svnmerge from
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r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines
Merged revisions 281911 via svnmerge from
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r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines
Ensure SSRC is changed when media source is changed to resolve audio delay.
This change causes the SSRC to change right before the channels are bridged,
which is what used to happen. It seems that fixes were made to attempt limiting
SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change.
There are two other control frames sent in ast_channel_bridge that probably
should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave
this change up to the discretion of resolving issue #17007.
For reference - old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540
(closes issue #17404)
Reported by: sdolloff
Patches:
bug17404.patch uploaded by jpeeler (license 325)
Tested by: sdolloff
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Leif Madsen [Wed, 11 Aug 2010 21:13:06 +0000 (21:13 +0000)]
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r281875 | lmadsen | 2010-08-11 16:12:13 -0500 (Wed, 11 Aug 2010) | 21 lines
Merged revisions 281873 via svnmerge from
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r281873 | lmadsen | 2010-08-11 16:09:47 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281819 via svnmerge from
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r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines
Add Danish support to say.conf.sample
(closes issue #17836)
Reported by: RoadKill
Patches:
say.conf.sample.patch.dk uploaded by RoadKill (license 933)
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Matthew Nicholson [Wed, 11 Aug 2010 21:12:25 +0000 (21:12 +0000)]
Merged revisions 281874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug 2010) | 10 lines
handle all possible responses to REFER requests
(closes issue #17486)
Reported by: davidw
Patches:
Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
Tested by: davidw
Review: https://reviewboard.asterisk.org/r/837/
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Richard Mudgett [Wed, 11 Aug 2010 20:38:57 +0000 (20:38 +0000)]
Merged revisions 281870 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) | 4 lines
Fix a call to analog_set_pulsedial() not setting 0 or 1 only.
* Also a couple minor tweaks.
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Leif Madsen [Wed, 11 Aug 2010 17:55:45 +0000 (17:55 +0000)]
Merged revisions 281764 via svnmerge from
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r281764 | lmadsen | 2010-08-11 12:54:56 -0500 (Wed, 11 Aug 2010) | 21 lines
Merged revisions 281763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281763 | lmadsen | 2010-08-11 12:54:09 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines
Allow say.conf to handle large numbers ending with multiple zeros.
(closes issue #17833)
Reported by: RoadKill
Patches:
say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933)
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Matthew Nicholson [Wed, 11 Aug 2010 17:29:16 +0000 (17:29 +0000)]
Merged revisions 281760 via svnmerge from
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r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug 2010) | 4 lines
Avoid a deadlock in add_header_max_forwards().
Related to r276951
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Tilghman Lesher [Wed, 11 Aug 2010 15:20:29 +0000 (15:20 +0000)]
Merged revisions 281723 via svnmerge from
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r281723 | tilghman | 2010-08-11 10:18:40 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) | 7 lines
Only set status TIMEOUT, if we have no digits.
(closes issue #15188)
Reported by: jcovert
Patches:
app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551)
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