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git.ipfire.org Git - thirdparty/asterisk.git/log
Jason Parker [Tue, 9 Aug 2011 15:51:38 +0000 (15:51 +0000)]
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r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line
Revert merge of r306999, due to merge conflict.
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Terry Wilson [Mon, 8 Aug 2011 22:59:45 +0000 (22:59 +0000)]
Merged revisions 331097 via svnmerge from
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r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines
Bump the AMI protocol version to 1.2
As a result of converting Unlink events that were missed in the AMI
1.1 update to Bridge events, the AMI protocol version is being incremented.
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Terry Wilson [Mon, 8 Aug 2011 21:16:25 +0000 (21:16 +0000)]
Merged revisions 331041 via svnmerge from
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r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines
Replace AMI Unlink events with Bridge events
A previous update converted some of the Link and Unlink events to
Bridge events, but a couple of Unlink events were missed. This patch
rectifies the situation.
(closes issues ASTERISK-17455)
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Kinsey Moore [Mon, 8 Aug 2011 20:54:14 +0000 (20:54 +0000)]
Merged revisions 331039 via svnmerge from
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r331039 | kmoore | 2011-08-08 15:53:30 -0500 (Mon, 08 Aug 2011) | 18 lines
Merged revisions 331038 via svnmerge from
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r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines
In-queue MOH stops after a periodic announcement
If the seek value is past the end of file when resuming G.722 MOH, MOH will
cease to function for the duration of the MOH session through all starts and
stops until saved state is cleared. Adjusting the code to guarantee a single
valid read (which is already assumed) fixes the bug.
(closes issue ASTERISK-18077)
Review: https://reviewboard.asterisk.org/r/1328/
Tested-by: Jonathan Rose <jrose@digium.com>
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Kinsey Moore [Mon, 8 Aug 2011 20:28:20 +0000 (20:28 +0000)]
Log queue member name when state_interface is set for ADDMEMBER and REMOVEMEMBER events
app_queue logs the events ADDMEMBER and REMOVEMEMBER with the agent field set
to the interface value rather than the membername value when a member is added
with a state_interface value set. However all other member related queue
events are logged with the membername when a state_interface is set. This
patch makes these fields optionally more consistent and correct.
(closes issue ASTERISK-14769)
Review: https://reviewboard.asterisk.org/r/1286
Patch-by: Jamuel Starkey
Tested-by: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331037
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Kinsey Moore [Mon, 8 Aug 2011 15:00:26 +0000 (15:00 +0000)]
app_queue: Add StateInterface to output of "queue show" and "QueueStatus"
This patch adds the state_interface of the queue member struct to the output
of "queue show" (CLI command) and "QueueStatus" (AMI action) when displaying
relevant queue member information. For the AMI event message the variable
StateInterface has been added.
(closes issue ASTERISK-18071)
Review: https://reviewboard.asterisk.org/r/1300/
Patch-by: Jamuel Starkey
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331000
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David Vossel [Fri, 5 Aug 2011 15:57:06 +0000 (15:57 +0000)]
Merged revisions 330940 via svnmerge from
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r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011) | 2 lines
The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly.
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Alexandr Anikin [Fri, 5 Aug 2011 08:47:59 +0000 (08:47 +0000)]
Merged revisions 330899 via svnmerge from
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r330899 | may | 2011-08-05 11:38:28 +0400 (Fri, 05 Aug 2011) | 11 lines
Merged revisions 330827 via svnmerge from
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r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines
change gk client behaivour on rrq/grq failures to setup timers
and next tries after timeout instead of complete failure in the ooh323
stack
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Terry Wilson [Thu, 4 Aug 2011 20:53:16 +0000 (20:53 +0000)]
Merged revisions 330844 via svnmerge from
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r330844 | twilson | 2011-08-04 15:51:23 -0500 (Thu, 04 Aug 2011) | 11 lines
Merged revisions 330843 via svnmerge from
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r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011) | 4 lines
Make libsrtp instructions more explicit when linking fails
(closes issue ASTERISK-18139)
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Kinsey Moore [Wed, 3 Aug 2011 15:16:25 +0000 (15:16 +0000)]
Merged revisions 330763 via svnmerge from
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r330763 | kmoore | 2011-08-03 10:15:26 -0500 (Wed, 03 Aug 2011) | 16 lines
Merged revisions 330762 via svnmerge from
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r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) | 9 lines
editing files in main/editline does not ensure rebuild of libedit.a
When editing a source file in main/editline, the build system does not rebuild
libedit.a and uses the already existing one instead. Adding a PHONY to
CHECK_SUBDIR fixes this problem.
(closes issue ASTERISK-16221)
Patch-by: Walter Doekes
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Kinsey Moore [Wed, 3 Aug 2011 13:40:22 +0000 (13:40 +0000)]
Merged revisions 330706 via svnmerge from
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r330706 | kmoore | 2011-08-03 08:39:06 -0500 (Wed, 03 Aug 2011) | 17 lines
Merged revisions 330705 via svnmerge from
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r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines
Call pickup broken for DAHDI channels when beginning with #
The call pickup feature did not work on DAHDI devices for anything other than
feature codes beginning with * since all feature codes in chan_dahdi were
originally hard-coded to begin with *. This patch is also applied to
chan_dahdi.c to fix this bug with radio modes.
(closes issue AST-621)
Review: https://reviewboard.asterisk.org/r/1336/
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Kevin P. Fleming [Tue, 2 Aug 2011 20:54:19 +0000 (20:54 +0000)]
Merged revisions 330649 via svnmerge from
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r330649 | kpfleming | 2011-08-02 15:52:44 -0500 (Tue, 02 Aug 2011) | 9 lines
Merged revisions 330648 via svnmerge from
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r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 Aug 2011) | 2 lines
Convert an error message to actually be helpful.
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David Vossel [Tue, 2 Aug 2011 16:19:32 +0000 (16:19 +0000)]
Merged revisions 330586 via svnmerge from
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r330586 | dvossel | 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) | 15 lines
Merged revisions 330581 via svnmerge from
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r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines
Fixes crash in chan_iax2.
Fixes crash in chan_iax2 resulting from an edge case in the
way control frames are queued during calltoken negotiation is complete.
(closes issue ASTERISK-17610)
Reported by: mgrobecker
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David Vossel [Tue, 2 Aug 2011 16:09:50 +0000 (16:09 +0000)]
Merged revisions 330579 via svnmerge from
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r330579 | dvossel | 2011-08-02 11:08:57 -0500 (Tue, 02 Aug 2011) | 9 lines
Merged revisions 330578 via svnmerge from
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r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) | 2 lines
Optimization to buffer initialization fix.
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David Vossel [Tue, 2 Aug 2011 16:04:34 +0000 (16:04 +0000)]
Merged revisions 330576 via svnmerge from
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r330576 | dvossel | 2011-08-02 10:55:36 -0500 (Tue, 02 Aug 2011) | 12 lines
Merged revisions 330575 via svnmerge from
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r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) | 5 lines
Fixes uninitialized string buffer in log message.
(closes issue ASTERISK-17200)
Reported by: lmadsen
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Jonathan Rose [Mon, 1 Aug 2011 21:22:08 +0000 (21:22 +0000)]
Blocked revisions 330514 via svnmerge
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r330514 | jrose | 2011-08-01 16:21:23 -0500 (Mon, 01 Aug 2011) | 14 lines
Blocked revisions 330510 via svnmerge
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r330510 | jrose | 2011-08-01 16:20:40 -0500 (Mon, 01 Aug 2011) | 8 lines
Blocked revisions 330505 via svnmerge
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r330505 | jrose | 2011-08-01 16:19:47 -0500 (Mon, 01 Aug 2011) | 3 lines
fixes reference leak pointed out by rmudgett in https://reviewboard.asterisk.org/r/1337/
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Jonathan Rose [Mon, 1 Aug 2011 21:11:24 +0000 (21:11 +0000)]
Blocked revisions 330492 via svnmerge
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r330492 | jrose | 2011-08-01 16:10:35 -0500 (Mon, 01 Aug 2011) | 24 lines
Blocked revisions 330491 via svnmerge
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r330491 | jrose | 2011-08-01 16:09:15 -0500 (Mon, 01 Aug 2011) | 18 lines
Blocked revisions 330490 via svnmerge
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r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines
Asterisk 18103 - Fix reload crash caused by destroying default parking lot
Default parking lot was being destroyed in reload and was not being rebuilt properly.
This patch keeps features.c reload from destroying the default parking lot in 1.6.2.
Bug was caused by a hasty backport which didn't test reload enough times to catch the
problem.
(Closes Issue ASTERISK-18103)
Reported by: 808blogger
Review: https://reviewboard.asterisk.org/r/1337/
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Kinsey Moore [Mon, 1 Aug 2011 15:24:21 +0000 (15:24 +0000)]
Merged revisions 330434 via svnmerge from
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r330434 | kmoore | 2011-08-01 10:23:29 -0500 (Mon, 01 Aug 2011) | 16 lines
Merged revisions 330433 via svnmerge from
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r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | 9 lines
Incorrect playback for Spanish in some circumstances
When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you
must use female pronunciation "1F". The function "say_date_with_format_es" does
not take this in account.
(closes ASTERISK-15016)
Patch-by: Luis Jimenez
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Richard Mudgett [Sun, 31 Jul 2011 00:19:11 +0000 (00:19 +0000)]
Fixed compiler warning and a couple prototype mismatches.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330379
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Richard Mudgett [Sun, 31 Jul 2011 00:05:55 +0000 (00:05 +0000)]
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r330369 | rmudgett | 2011-07-30 18:57:56 -0500 (Sat, 30 Jul 2011) | 11 lines
Merged revisions 330368 via svnmerge from
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r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines
Remove some redundant locking code in ast_do_masquerade().
Also updated some comments.
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Gregory Nietsky [Sat, 30 Jul 2011 15:54:23 +0000 (15:54 +0000)]
Merged revisions 330312 via svnmerge from
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r330312 | irroot | 2011-07-30 17:34:41 +0200 (Sat, 30 Jul 2011) | 15 lines
Merged revisions 330311 via svnmerge from
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r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines
prevent double masqurading channels when one is been hung up and deadlock avoidance is used.
There is a race condition in ast_do_masquerade / ast_hangup (at least)
Reported by me signed off by schmidts with input from David Vossel
Review: https://reviewboard.asterisk.org/r/1323/
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Russell Bryant [Fri, 29 Jul 2011 19:34:36 +0000 (19:34 +0000)]
astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER.
There is a fairly common pattern making its way through the code base where we
put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER.
The purpose is so that it can be passed into the object hash function.
However, this really seems like a hack and potentially error prone. This patch
is a first stab at approach to avoid having to do that.
It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these
situations. Then, the hash function can know whether it was given an object or
some custom data to hash.
The patch also changes some uses of ao2_find() for iax2_user and iax2_peer
objects to reflect how OBJ_KEY would be used.
So long, and thanks for all the fish.
Review: https://reviewboard.asterisk.org/r/1184/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330273
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Sean Bright [Fri, 29 Jul 2011 17:20:20 +0000 (17:20 +0000)]
Merged revisions 330217 via svnmerge from
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r330217 | seanbright | 2011-07-29 13:19:42 -0400 (Fri, 29 Jul 2011) | 9 lines
Merged revisions 330213 via svnmerge from
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r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) | 2 lines
Correct the check for O_RDONLY.
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Sean Bright [Fri, 29 Jul 2011 17:02:11 +0000 (17:02 +0000)]
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r330204 | seanbright | 2011-07-29 12:58:40 -0400 (Fri, 29 Jul 2011) | 9 lines
Merged revisions 330203 via svnmerge from
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r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) | 2 lines
Only write to wav files that were opened to be written to.
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Paul Belanger [Fri, 29 Jul 2011 05:27:22 +0000 (05:27 +0000)]
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r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul 2011) | 4 lines
Fix typo pointed out on #asterisk
Thanks notten
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Terry Wilson [Thu, 28 Jul 2011 21:46:27 +0000 (21:46 +0000)]
Merged revisions 330108 via svnmerge from
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r330108 | twilson | 2011-07-28 16:44:31 -0500 (Thu, 28 Jul 2011) | 9 lines
Merged revisions 330107 via svnmerge from
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r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011) | 2 lines
Make console colors work for TERM=xterm-256color
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Richard Mudgett [Thu, 28 Jul 2011 17:16:07 +0000 (17:16 +0000)]
Merged revisions 330051 via svnmerge from
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r330051 | rmudgett | 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) | 29 lines
Merged revisions 330050 via svnmerge from
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r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines
Merged revisions 330033 from
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r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines
Datacalls with B410P fail.
Incoming and outgoing call legs of a data call are using different
formats: a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation is bad
for data calls.
* Make incoming call that does not explicitly specify u-law or a-law use
the DAHDI channel's default law. The outgoing call always uses the
default law from the DAHDI channel.
(closes issue ABE-2800)
Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett
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Jason Parker [Thu, 28 Jul 2011 15:46:16 +0000 (15:46 +0000)]
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r329995 | qwell | 2011-07-28 10:45:49 -0500 (Thu, 28 Jul 2011) | 13 lines
Merged revisions 329994 via svnmerge from
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r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | 6 lines
Fix a SIP transfer deadlock.
The locking in this function is very scary. There are like 6 structs involved.
(closes issue AST-470)
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Matthew Nicholson [Thu, 28 Jul 2011 15:30:30 +0000 (15:30 +0000)]
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r329992 | mnicholson | 2011-07-28 10:28:21 -0500 (Thu, 28 Jul 2011) | 13 lines
Merged revisions 329991 via svnmerge from
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r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul 2011) | 6 lines
check for CONFIG_STATUS_FILE_INVALID when loading the res_fax config file
Patch by: tzafrir
Reported by: tzafrir
(closes issue ASTERISK-18161)
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Sean Bright [Thu, 28 Jul 2011 13:04:33 +0000 (13:04 +0000)]
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r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu, 28 Jul 2011) | 4 lines
The default conf-usermenu says that '8' can be used to leave the conference, so
put that in the sample user menu. '5' is supposed to extend the conference, but
there doesn't appear to be a concept of that in the menu actions.
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Sean Bright [Thu, 28 Jul 2011 12:44:51 +0000 (12:44 +0000)]
Merged revisions 329950 via svnmerge from
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r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul 2011) | 1 line
Correct the spelling of 'conference.'
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Sean Bright [Thu, 28 Jul 2011 11:36:12 +0000 (11:36 +0000)]
Merged revisions 329896 via svnmerge from
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r329896 | seanbright | 2011-07-28 07:35:27 -0400 (Thu, 28 Jul 2011) | 9 lines
Merged revisions 329895 via svnmerge from
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r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) | 2 lines
Make the output of Externhost in 'sip show settings' more consistent.
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Jonathan Rose [Wed, 27 Jul 2011 21:22:12 +0000 (21:22 +0000)]
reverting 329840 due to failing tests. Going to change this feature to be purely optional.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329856
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Jonathan Rose [Wed, 27 Jul 2011 20:42:18 +0000 (20:42 +0000)]
Adds cdr logging of calls resulting in CONGESTION
Applies a patch made a long time ago by alecdavis which adds a CDR feature for logging
calls that failed due to congestion.
(closes issue #15907)
Reported by: alecdavis
Patches:
cdr_congestion.diff.txt uploaded by alecdavis (license #5546)
Review: https://reviewboard.asterisk.org/r/454/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329835
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Sean Bright [Wed, 27 Jul 2011 19:19:46 +0000 (19:19 +0000)]
Merged revisions 329771 via svnmerge from
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r329771 | seanbright | 2011-07-27 15:18:47 -0400 (Wed, 27 Jul 2011) | 15 lines
Merged revisions 329767 via svnmerge from
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r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul 2011) | 8 lines
Explicitly sort the module list so that the menuselect lists are sorted.
(closes ASTERISK-18141)
Reported by: Richard Miller
Patches:
sort-order.diff uploaded by seanbright (License #5060)
Tested by: leifmadsen
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Jonathan Rose [Wed, 27 Jul 2011 18:12:14 +0000 (18:12 +0000)]
Merged revisions 329710 via svnmerge from
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r329710 | jrose | 2011-07-27 13:11:07 -0500 (Wed, 27 Jul 2011) | 14 lines
Merged revisions 329709 via svnmerge from
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r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) | 8 lines
Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf
(closes issue ASTERISK-16263)
Reported by: richardf
Patches:
nz-indications.patch uploaded by richardf (License #6015)
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Sean Bright [Wed, 27 Jul 2011 15:26:31 +0000 (15:26 +0000)]
Merged revisions 329670 via svnmerge from
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r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul 2011) | 2 lines
Sort the module list so that 'module show' is alphabetical.
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Tilghman Lesher [Wed, 27 Jul 2011 04:27:19 +0000 (04:27 +0000)]
Merged revisions 329614 via svnmerge from
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r329614 | tilghman | 2011-07-26 23:25:26 -0500 (Tue, 26 Jul 2011) | 13 lines
Merged revisions 329613 via svnmerge from
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r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011) | 6 lines
Duration and billsec are swapped in high resolution time.
Closes ASTERISK-18024
Patches:
20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
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Jonathan Rose [Tue, 26 Jul 2011 14:27:31 +0000 (14:27 +0000)]
Merged revisions 329538 via svnmerge from
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r329538 | jrose | 2011-07-26 09:19:34 -0500 (Tue, 26 Jul 2011) | 11 lines
Merged revisions 329529 via svnmerge from
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r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | 5 lines
Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same
prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more
appropriate anyway.
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Jonathan Rose [Tue, 26 Jul 2011 14:17:13 +0000 (14:17 +0000)]
Merged revisions 329528 via svnmerge from
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r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines
Merged revisions 329527 via svnmerge from
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r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines
Fixes some voicemail forwarding behavior based around prepend mode.
Formerly, prepend forwarding would have the user record a message with no useful prompt
and an expectation for the user to push a button on the phone when finished recording.
If a length of silence was detected instead, the recording would be canceled and the user
would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording
would also bug out in the sense that they would write over the original message and get
sent to the recipient regardless of whether they timed out or were accepted. This patch
fixes this issue and adds a prompt which will be played after a timeout informing the
user that they needed to press a button. Currently, the sound files that we have are
somewhat inadquate for this, so after the call we simply have Allison say "Please try
again. Then press pound." which actually relies on two separate sound files. Just one
would be more appropriate.
reporter: Vlad Povorozniuc
Review: https://reviewboard.asterisk.org/r/1327/
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Paul Belanger [Mon, 25 Jul 2011 19:57:27 +0000 (19:57 +0000)]
Merged revisions 329472 via svnmerge from
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r329472 | pabelanger | 2011-07-25 15:55:33 -0400 (Mon, 25 Jul 2011) | 9 lines
Merged revisions 329471 via svnmerge from
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r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul 2011) | 2 lines
Decrease verbose messages to debug, to help clean up CLI.
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Gregory Nietsky [Mon, 25 Jul 2011 14:07:01 +0000 (14:07 +0000)]
dsp_process was enhanced to work with alaw and ulaw in addition to slin.
noticed that some functions could be refactored here it is.
Reported by: irroot
Tested by: irroot, mnicholson
Review: https://reviewboard.asterisk.org/r/1304/
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Gregory Nietsky [Mon, 25 Jul 2011 09:39:54 +0000 (09:39 +0000)]
Remove lastmsgssent from sip it has not been working since 1.6
Clean up the return values to be consistant not currently used
Add doxygen returns
MWI Event is sent on Register
(closes issue ASTERISK-17866)
Reported by: one47
Tested by: irroot, mvanbaak
Review: https://reviewboard.asterisk.org/r/1172/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329391
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Richard Mudgett [Fri, 22 Jul 2011 21:15:28 +0000 (21:15 +0000)]
Merged revisions 329334 via svnmerge from
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r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011) | 1 line
Make use less redundant loop construct for iterating over hints.
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Richard Mudgett [Fri, 22 Jul 2011 20:46:36 +0000 (20:46 +0000)]
Merged revisions 329331 via svnmerge from
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r329331 | rmudgett | 2011-07-22 15:43:07 -0500 (Fri, 22 Jul 2011) | 55 lines
Merged revisions 329299 via svnmerge from
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r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines
Deadlocks dealing with dialplan hints during reload.
There are two remaining different deadlocks reported dealing with dialplan
hints.
The deadlock in ASTERISK-17666 is caused by invalid locking order in
ast_remove_hint(). The hints container must be locked before the hint
object.
The deadlock in ASTERISK-17760 is caused by a catch-22 situation in
handle_statechange(). The deadlock is caused by not having the conlock
before calling the watcher callbacks. Unfortunately, having that lock
causes a different deadlock as reported in ASTERISK-16961.
* Fixed ast_remove_hint() locking order.
* Made handle_statechange() no longer call the watcher callbacks holding
any locks that matter.
* Made hint ao2 destructor do the watcher callbacks for extension
deactivation to guarantee that they get called.
* Fixed hint reference leak in ast_add_hint() if the callback container
constructor failed.
* Fixed hint reference leak in complete_core_show_hint() for every hint it
found for CLI tab completion.
* Adjusted locking in ast_merge_contexts_and_delete() for safety.
* Added context_merge_lock to prevent ast_merge_contexts_and_delete() and
handle_statechange() from interfering with each other.
* Fixed ast_change_hint() not taking into account that the extension is
used for the hash key.
(closes issue ASTERISK-17666)
Reported by: irroot
Tested by: irroot
JIRA SWP-3318
(closes issue ASTERISK-17760)
Reported by: Byron Clark
Tested by: irroot
JIRA SWP-3393
Review: https://reviewboard.asterisk.org/r/1313/
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Russell Bryant [Thu, 21 Jul 2011 20:26:44 +0000 (20:26 +0000)]
Merged revisions 329257 via svnmerge from
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r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines
s/1.10/10.0/
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Richard Mudgett [Thu, 21 Jul 2011 18:06:47 +0000 (18:06 +0000)]
Merged revisions 329204 via svnmerge from
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r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines
Merged revisions 329203 via svnmerge from
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r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines
Document parkinglot in chan_dahdi.conf.sample.
* Document existing feature in chan_dahdi.conf.sample.
* Remove some dead code related to the parkinglot option.
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Richard Mudgett [Thu, 21 Jul 2011 17:33:06 +0000 (17:33 +0000)]
Merged revisions 329200 via svnmerge from
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r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines
Merged revisions 329199 via svnmerge from
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r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines
Update PickupChan documentation.
The PickupChan uses the ampersand as the argument separator.
Was documented as:
PickupChan(channel[,channel2[,...][,options]])
Fixed documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144)
Reported by: Erik Smith
Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
Tested by: Erik Smith
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Richard Mudgett [Thu, 21 Jul 2011 16:59:38 +0000 (16:59 +0000)]
Merged revisions 329145 via svnmerge from
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r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines
Merged revisions 329144 via svnmerge from
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r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines
Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked!
This appears to be a leftover from when ast_channel was converted to ao2
objects.
Simply removed the extraneous unlock.
(closes issue ASTERISK-17772)
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Jason Parker [Thu, 21 Jul 2011 16:22:58 +0000 (16:22 +0000)]
Fix UPGRADE.txt files for Asterisk 10.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329130
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Jason Parker [Thu, 21 Jul 2011 16:14:44 +0000 (16:14 +0000)]
Remove another 2.0 property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329106
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Russell Bryant [Thu, 21 Jul 2011 16:05:33 +0000 (16:05 +0000)]
Fix merge properties to reflect Asterisk 10 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329105
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Paul Belanger [Wed, 20 Jul 2011 21:31:29 +0000 (21:31 +0000)]
Merged revisions 329055 via svnmerge from
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r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines
Merged revisions 329027 via svnmerge from
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r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines
Asterisk now requires libpri 1.4.11+ for PRI support.
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Terry Wilson [Wed, 20 Jul 2011 20:19:16 +0000 (20:19 +0000)]
Merged revisions 328992 via svnmerge from
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r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines
Merged revisions 328987 via svnmerge from
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r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines
We can't guarantee an eth0 is present
FreeBSD test fails on this case presumably because there is no eth0 on the test
machine. Better to just remove this test for now.
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Kinsey Moore [Wed, 20 Jul 2011 19:03:17 +0000 (19:03 +0000)]
Merged revisions 328936 via svnmerge from
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r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
Merged revisions 328935 via svnmerge from
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r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
Inband DTMF regression
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
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Kevin P. Fleming [Tue, 19 Jul 2011 21:32:54 +0000 (21:32 +0000)]
Merged revisions 328879 via svnmerge from
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r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines
Merged revisions 328878 via svnmerge from
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r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines
Revert partial attempt at handling pathnames with spaces.
Revision 299794 attempted to improve the build system to be able to handle
pathnames (primarily DESTDIR) with spaces in them, since this is common on
some platforms (including Mac OSX). Unfortunately, the changes were incomplete
and did not actually provide the desired behavior, and as a side effect the
functionality that ensured that stale headers in the Asterisk 'include' directory
were removed got broken. In addition, the check for stale (and possibly
incompatible) modules in the Asterisk 'modules' directory also got broken, and
would never report any stale modules. Users upgrading to this version or later
versions would then see unexpected module load errors.
Since there are few users who actually want to install Asterisk into paths
that contain spaces, and a proper fix for the build system would take many hours,
the best solution for now is to just revert the partial solution.
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Kevin P. Fleming [Tue, 19 Jul 2011 21:32:35 +0000 (21:32 +0000)]
Edit the merge properties to match their names.
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Russell Bryant [Tue, 19 Jul 2011 21:21:53 +0000 (21:21 +0000)]
Fix properties after twilson's change so merges still work
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Kinsey Moore [Tue, 19 Jul 2011 18:07:22 +0000 (18:07 +0000)]
Merged revisions 328824 via svnmerge from
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r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328823 via svnmerge from
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r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
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Kinsey Moore [Tue, 19 Jul 2011 15:49:55 +0000 (15:49 +0000)]
Merged revisions 328771 via svnmerge from
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r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328770 via svnmerge from
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r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
MeetMe requests a PIN twice in some circumstances
If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
options, MeetMe will ask for the PIN two times: once for creating the
conference and once for entering the conference. This behavior was introduced
in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
controlling PIN entry for joining a conference.
(closes AST-601)
Review: https://reviewboard.asterisk.org/r/1305/
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Terry Wilson [Tue, 19 Jul 2011 02:00:56 +0000 (02:00 +0000)]
Merged revisions 328717 via svnmerge from
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r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines
Merged revisions 328716 via svnmerge from
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r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines
Make AST_LIST_REMOVE safer
AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This
commit also adds linked list unit tests.
Review: https://reviewboard.asterisk.org/r/1321/
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Mark Murawki [Mon, 18 Jul 2011 20:51:47 +0000 (20:51 +0000)]
Merged revisions 328664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
app_dial may double free a channel datastore
When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917)
Reported by: Mark Murawski
Tested by: Mark Murawski
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328665
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Mark Murawki [Mon, 18 Jul 2011 12:58:02 +0000 (12:58 +0000)]
Merged revisions 328611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
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Mark Murawki [Mon, 18 Jul 2011 12:54:29 +0000 (12:54 +0000)]
Merged revisions 328609 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines
Fixed invalid read and null pointer deref on asterisk shutdown.
In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash.
(closes issue ASTERISK-17927)
Reported by: Mark Murawski
Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher
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Tilghman Lesher [Mon, 18 Jul 2011 07:12:22 +0000 (07:12 +0000)]
Merged revisions 328541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328541 | tilghman | 2011-07-18 02:11:26 -0500 (Mon, 18 Jul 2011) | 9 lines
Merged revisions 328540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines
Typo
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Alexandr Anikin [Fri, 15 Jul 2011 21:41:46 +0000 (21:41 +0000)]
Merged revisions 328428-328429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines
Merged revisions 328427 via svnmerge from
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r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines
small gk processing fixes:
- decrease for 1 second registration ttl for very low expirations (some
providers expire few earlier than TTL)
- delete rrq and registration expire timers on URQ received as we make
new registration.
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r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines
delete unproperly changed svn props
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Leif Madsen [Fri, 15 Jul 2011 21:19:08 +0000 (21:19 +0000)]
Merged revisions 328451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line
Build app_macro by default because things depend on it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328459
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Leif Madsen [Fri, 15 Jul 2011 21:01:41 +0000 (21:01 +0000)]
Merged revisions 328448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines
Update UPGRADE.txt and CHANGES files.
Update documentation files stating that deprecated modules are no longer built by default.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328449
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Damien Wedhorn [Fri, 15 Jul 2011 08:19:46 +0000 (08:19 +0000)]
Add SLA to skinny.
Adds sublines to skinny lines. Each subline can be attached to an
SLA station/trunk combo. Includes the following functionality:
Callid is persistent for both in/out calls on all skinny devices.
Can join, hold, resume.
All sublines appear under a single line button.
See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc.
(closes issue ASTERISK-17947)
Review: https://reviewboard.asterisk.org/r/1239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328381
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Richard Mudgett [Fri, 15 Jul 2011 00:23:14 +0000 (00:23 +0000)]
Merged revisions 328329 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines
Make hint watcher callback take const strings for context and exten parameters.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328344
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Richard Mudgett [Thu, 14 Jul 2011 23:34:43 +0000 (23:34 +0000)]
Merged revisions 328317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines
Merged revisions 328302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines
Missing SIP pvt and channel unlock in sip_set_rtp_peer().
Regression introduced by -r326144.
Add missing SIP pvt and channel unlock in sip_set_rtp_peer().
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Leif Madsen [Thu, 14 Jul 2011 20:28:54 +0000 (20:28 +0000)]
Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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Jonathan Rose [Thu, 14 Jul 2011 19:56:19 +0000 (19:56 +0000)]
Merged revisions 328207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines
Merged revisions 328205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines
Monitor application arguments requirements fixed.
Monitor was requiring options in spite of no individual option on Monitor being required.
Review: https://reviewboard.asterisk.org/r/1320/
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Matthew Nicholson [Thu, 14 Jul 2011 17:47:40 +0000 (17:47 +0000)]
Merged revisions 328162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines
tune the v21 preamble detector to properly detect the desired sequence of hits
and misses
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328163
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David Vossel [Wed, 13 Jul 2011 22:10:26 +0000 (22:10 +0000)]
Merged revisions 328120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines
Preserve sample rate quality of wideband mixmonitor recordings.
MixMonitor has the ability to record in any file format Asterisk supports,
but the quality of wideband audio is not preserved. This is because
regardless of the sample rate the call is being recorded in, the audio
is always downsampled to 8khz and then upsampled to whatever wideband
format it is being written as. This patch resolves this by requesting
the audio from the audiohook in the signed linear format closest to the
sample rate of the format we are writing. This fix is only possible for
Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
audio.
Review: https://reviewboard.asterisk.org/r/1314/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328121
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Leif Madsen [Wed, 13 Jul 2011 21:06:23 +0000 (21:06 +0000)]
Add UPGRADE-1.10.txt file from UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328079
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Russell Bryant [Wed, 13 Jul 2011 20:40:19 +0000 (20:40 +0000)]
set 1.10 merge properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328076
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Russell Bryant [Wed, 13 Jul 2011 20:37:13 +0000 (20:37 +0000)]
remove 1.8 merge properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328075
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Richard Mudgett [Wed, 13 Jul 2011 18:47:16 +0000 (18:47 +0000)]
Merged revisions 328014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) | 1 line
Add ATXFER_NULL_TECH note in features.conf.sample.
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Kevin P. Fleming [Tue, 12 Jul 2011 23:02:31 +0000 (23:02 +0000)]
Merged revisions 327950 via svnmerge from
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r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines
Correct double-free situation in manager output processing.
The process_output() function calls ast_str_append() and xml_translate() on its
'out' parameter, which is a pointer to an ast_str buffer. If either of these
functions need to reallocate the ast_str so it will have more space, they will
free the existing buffer and allocate a new one, returning the address of the
new one. However, because process_output only receives a pointer to the ast_str,
not a pointer to its caller's variable holding the pointer, if the original
ast_str is freed, the caller will not know, and will continue to use it (and
later attempt to free it).
(reported by jkroon on #asterisk-dev)
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Matthew Nicholson [Tue, 12 Jul 2011 20:08:04 +0000 (20:08 +0000)]
Merged revisions 327890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) | 2 lines
search in the current context for 'a' and 'o' instead of 'default'
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Jason Parker [Tue, 12 Jul 2011 19:39:04 +0000 (19:39 +0000)]
Merged revisions 327888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | 1 line
Fix uninstall target, so that modules dir gets cleared again.
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Matthew Jordan [Tue, 12 Jul 2011 19:18:08 +0000 (19:18 +0000)]
Merged revisions 327852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines
Added additional checks for mailbox / password beginning with '*' character
A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.
(closes issue ASTERISK-17443)
Reported by: Kevin Scott Adams
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1316/
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Tilghman Lesher [Tue, 12 Jul 2011 15:38:47 +0000 (15:38 +0000)]
Merged revisions 327793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines
Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability.
The problem with using 'echo -n' is that it is not portable. While BSD systems
required that the '-n' option be removed and interpreted, System V required
that all strings should be echoed with no interpretation of options. This
fundamental difference of behavior means that it is never possible to use the
'-n' flag to echo in tests which are meant to be portable.
In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the
System V semantics of the command, and thus the SHELL test failed on that
platform.
http://pubs.opengroup.org/onlinepubs/
009695399 /utilities/echo.html#tag_04_41_16
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Matthew Nicholson [Tue, 12 Jul 2011 15:23:24 +0000 (15:23 +0000)]
do v21 detection instead of CED detection for the fax gateway
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769
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David Vossel [Tue, 12 Jul 2011 14:55:51 +0000 (14:55 +0000)]
Send video update frame to new video source in follow_talker correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327749
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Kinsey Moore [Tue, 12 Jul 2011 14:40:16 +0000 (14:40 +0000)]
Segfault on shutdown when confbridge is active
When undergoing a shutdown and channels are kicked out of a bridge, a segfault
occurs because ConfBridge tries to play sounds on the bridge after the
underlying channels have been blown away due to the shutdown.
(closes ASTERISK-18040)
Review: https://reviewboard.asterisk.org/r/1283/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327748
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Matthew Nicholson [Mon, 11 Jul 2011 20:06:28 +0000 (20:06 +0000)]
use printf instead of echo -n in substitution test
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327684
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Terry Wilson [Mon, 11 Jul 2011 19:49:35 +0000 (19:49 +0000)]
Merged revisions 327682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines
Update chan_gtalk to work with changed GMail-based calls
The messages sent by the GMail client have changed, but include the
old-style messages as well. This patch checks for this case and
uses the old-style offer.
(closes issue ASTERISK-18084)
Review: https://reviewboard.asterisk.org/r/1312/
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David Vossel [Mon, 11 Jul 2011 18:44:06 +0000 (18:44 +0000)]
Updates follow_talker video_mode in confbridge application.
follow_talker mode originally echoed the same video stream
to all participants. As the primary talker switched around, the
video stream would result in the talker seeing themselves. Now
the primary talker sees the last person who was talking rather than
themselves.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327640
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Matthew Nicholson [Mon, 11 Jul 2011 17:23:54 +0000 (17:23 +0000)]
renamed fax_gateway_send_ced() to fax_gateway_request_t38()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327598
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Matthew Nicholson [Mon, 11 Jul 2011 16:27:08 +0000 (16:27 +0000)]
actually do something with the ced timeout, also added more debug output
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327570
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Matthew Nicholson [Mon, 11 Jul 2011 14:13:24 +0000 (14:13 +0000)]
write silence on the channel during t.38 negotiation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327514
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Matthew Nicholson [Mon, 11 Jul 2011 13:55:28 +0000 (13:55 +0000)]
Merged revisions 327512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul 2011) | 2 lines
reset our buffer each iteration when doing variable substitution
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Matthew Nicholson [Mon, 11 Jul 2011 13:29:13 +0000 (13:29 +0000)]
Delay sending an CED tone generated T.38 reinvite to give the CED tone
generating party time to send its own T.38 reinvite.
Also don't forward frames through the gateway if we are negotiating T.38.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327511
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Matthew Nicholson [Mon, 11 Jul 2011 12:58:50 +0000 (12:58 +0000)]
fixed wording in a comment
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327469
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Tzafrir Cohen [Mon, 11 Jul 2011 10:57:26 +0000 (10:57 +0000)]
Merged revisions 327411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | 5 lines
fix building the Debian armhf (HardFloat) port
Fixes http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=
1309935385
(Missing pthreads)
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Alexandr Anikin [Sun, 10 Jul 2011 01:37:58 +0000 (01:37 +0000)]
Full T.38 handshaking and fax detection
Add full t.38 handshaking for OOH323 that are required for newest T.38
gateway codes.
Add fax detection (cng tone, t38) and dialplan redirection to fax ext on
fax event detected.
Add OOH323() function to set/get t38support and faxdetect parameters.
(closes issue ASTERISK-17754)
Reported by: irroot
Patches:
ooh323_faxdetect.patch uploaded by irroot (license 52)
issue19183-final.patch uploaded by may213 (license 454)
Tested by: may213, irroot
Review: https://reviewboard.asterisk.org/r/1174/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327359
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Jason Parker [Fri, 8 Jul 2011 22:25:14 +0000 (22:25 +0000)]
Add .o files to svn:ignore property, since it's only ignored if locally configured to do so.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327246
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Richard Mudgett [Fri, 8 Jul 2011 21:43:49 +0000 (21:43 +0000)]
Merged revisions 327211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
INVITE 403 Forbidden response always retransmits the maximum times.
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
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