Tilghman Lesher [Mon, 20 Apr 2009 22:03:08 +0000 (22:03 +0000)]
Blocked revisions 189537 via svnmerge
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r189537 | tilghman | 2009-04-20 17:02:16 -0500 (Mon, 20 Apr 2009) | 11 lines
Add a workaround for func_odbc/ARRAY() for problems that occur with certain special characters.
In certain cases, due to the way Set() works in 1.4, values may not get set
properly. This is a workaround for 1.4 only that corrects for these issues,
without making func_odbc more difficult to use properly.
(closes issue #14614)
Reported by: wdoekes
Patches:
20090309__bug14614__2.diff.txt uploaded by tilghman (license 14)
double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff uploaded by wdoekes (license 717)
Tested by: wdoekes, tilghman
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Terry Wilson [Mon, 20 Apr 2009 21:29:29 +0000 (21:29 +0000)]
Merged revisions 189465 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) | 2 lines
Update CDR appropriately when AST_CAUSE_NO_ANSWER is set
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Terry Wilson [Mon, 20 Apr 2009 21:24:34 +0000 (21:24 +0000)]
Merged revisions 189463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009) | 2 lines
Don't treat a NOANSWER like a CHANUNAVAIL
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Sean Bright [Mon, 20 Apr 2009 21:09:59 +0000 (21:09 +0000)]
Merged revisions 189462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines
Properly handle @s within hints in AEL.
AEL was not handling the case of a device hint containing an @ symbol, which
caused parking hints (e.g. hint(park:exten@context)) to error out the parser.
This patch makes AEL treat the @ the same way it treats colon and ampersand
now, meaning the characters are included in verbatim.
(closes issue #14941)
Reported by: bpgoldsb
Patches:
bug14941.patch uploaded by seanbright (license 71)
Tested by: bpgoldsb
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Doug Bailey [Mon, 20 Apr 2009 19:28:16 +0000 (19:28 +0000)]
Merged revisions 189391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) | 4 lines
Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results.
Change test of mmap results to compare against MAP_FAILED
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Joshua Colp [Mon, 20 Apr 2009 17:05:15 +0000 (17:05 +0000)]
Fix a bug with non-UDP connections that caused dialogs to not get freed.
This issue crept up because of a reference count issue on non-UDP based dialogs.
The dialog reference count was increased when transmitting a packet reliably but never
decreased. This caused the dialog structure to hang around despite being unlinked from
the dialogs container.
(closes issue #14919)
Reported by: vrban
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189350
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Mark Michelson [Mon, 20 Apr 2009 14:05:27 +0000 (14:05 +0000)]
Merged revisions 189277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines
Move the check for chan->fdno == -1 to after the zombie/hangup check.
Many users were finding that their hung up channels were staying up and
causing 100% CPU usage.
(issue #14723)
Reported by: seadweller
Patches:
14723_1-4-tip.patch uploaded by mmichelson (license 60)
Tested by: falves11, bamby
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David Vossel [Sat, 18 Apr 2009 01:28:45 +0000 (01:28 +0000)]
Merged revisions 189203 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines
Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
(closes issue #14091)
Reported by: evandro
Patches:
autologoff.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/225/
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Richard Mudgett [Fri, 17 Apr 2009 21:48:10 +0000 (21:48 +0000)]
Merged revisions 188833,189134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines
Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone.
JIRA ABE-1835
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r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
Modifed/added some debug messages.
JIRA ABE-1835
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Mark Michelson [Fri, 17 Apr 2009 20:20:23 +0000 (20:20 +0000)]
Prevent a crash when SIP blonde transferring an unbridged call.
If one attempts to use the attended transfer button on a SIP phone
to transfer an unbridged call (such as a call to an IVR) but hangs
up while the target of the transfer is still ringing, we need to not
crash.
The problem was that ast_hangup was called from outside the channel
thread.
AST-211
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189097
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Sean Bright [Fri, 17 Apr 2009 19:36:38 +0000 (19:36 +0000)]
Fix copy/paste error with 'transmit silence' flag.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189077
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Matthew Nicholson [Fri, 17 Apr 2009 15:44:18 +0000 (15:44 +0000)]
Merged revisions 189009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr 2009) | 5 lines
Make Busy() application set the CDR disposition to BUSY.
(closes issue #14306)
Reported by: cristiandimache
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Joshua Colp [Fri, 17 Apr 2009 14:44:56 +0000 (14:44 +0000)]
Merged revisions 188946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
Fix a bug where a value used to create the channel name was bogus.
This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.
(closes issue #14256)
Reported by: Nick_Lewis
Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
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Mark Michelson [Fri, 17 Apr 2009 14:33:50 +0000 (14:33 +0000)]
Fix a spacing issue that I claimed I would when I committed this code.
Nothing major though.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188942
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Joshua Colp [Fri, 17 Apr 2009 14:26:53 +0000 (14:26 +0000)]
Merged revisions 188937 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines
Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been.
(issue AST-210)
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Mark Michelson [Fri, 17 Apr 2009 13:29:33 +0000 (13:29 +0000)]
Several fixes to the extenpatternmatchnew logic.
1. Differentiate between literal characters in an extension
and characters that should be treated as a pattern match. Prior to
these fixes, an extension such as NNN would be treated as a pattern,
rather than a literal string of N's.
2. Fixed the logic used when matching an extension with a bracketed
expression, such as 2[5-7]6.
3. Removed all areas of code that were executed when NOT_NOW was
#defined. The code in these areas had the potential to crash, for
one thing, and the actual intent of these blocks seemed counterproductive.
4. Fixed many many coding guidelines problems I encountered while looking
through the corresponding code.
5. Added failure cases and warning messages for when duplicate extensions
are encountered.
6. Miscellaneous fixes to incorrect or redundant statements.
(closes issue #14615)
Reported by: steinwej
Tested by: mmichelson
Review: http://reviewboard.digium.com/r/194/
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Tilghman Lesher [Thu, 16 Apr 2009 21:57:37 +0000 (21:57 +0000)]
Merged revisions 188835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines
Only update realtime, if global option rtupdate != false
(closes issue #14885)
Reported by: deepesh
Patches:
20090413__bug14885.diff.txt uploaded by tilghman (license 14)
Tested by: deepesh
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Tilghman Lesher [Thu, 16 Apr 2009 21:03:31 +0000 (21:03 +0000)]
Merged revisions 188773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) | 4 lines
Umask should not be exported into global namespace.
(closes issue #14912)
Reported by: jcapp
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David Vossel [Thu, 16 Apr 2009 19:30:23 +0000 (19:30 +0000)]
SIP state notify reorganization
What I've done here is simply break up how a state NOTIFY is built. Originally both the XML and sip header information were built within the same function. While this does work, it does not allow for the creation of multipart/related message bodies that can contain multiple XML entries with only one sip header. Now a separate function builds the XML for each notify. This patch also makes maintaining and modifying state notifications in the future much less of a pain.
Review: http://reviewboard.digium.com/r/224/
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Joshua Colp [Thu, 16 Apr 2009 13:42:52 +0000 (13:42 +0000)]
Fix a bug with the dahdi_setoption callback in chan_dahdi.
This function incorrectly reported success even if the option was
unsupported. This was exposed by the options to change the underlying
channel format. The function now returns a failure if the option
is unsupported.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188705
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David Vossel [Wed, 15 Apr 2009 22:10:04 +0000 (22:10 +0000)]
Merged revisions 188646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines
National prefix inserted even when caller ID not available
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
(closes issue #13207)
Reported by: shawkris
Patches:
national_prefix.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/220/
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Mark Michelson [Wed, 15 Apr 2009 20:17:33 +0000 (20:17 +0000)]
Merged revisions 188582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines
Update ast_readvideo_callback to match ast_readaudio_callback.
This fixes potential refcount errors that may occur on ast_filestreams.
AST-208
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Mark Michelson [Wed, 15 Apr 2009 15:24:50 +0000 (15:24 +0000)]
Make the cancellation of the dial timeout on a call forward optional.
This introduces the 'z' option to app_dial. With it set, a call forward
will cancel any timeout originally set for this instance of the Dial
application.
AST-207
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Jeff Peeler [Wed, 15 Apr 2009 14:57:20 +0000 (14:57 +0000)]
Don't try to do anything in pri_check_restart with service messages unless
libpri supports it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188515
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Mark Michelson [Tue, 14 Apr 2009 23:28:13 +0000 (23:28 +0000)]
Fix a couple of queue member reference leaks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188470
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Joshua Colp [Tue, 14 Apr 2009 17:40:50 +0000 (17:40 +0000)]
Fix an incorrect clock rate when sending T140 text.
(closes issue #14029)
Reported by: epicac
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188413
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Jeff Peeler [Tue, 14 Apr 2009 16:49:12 +0000 (16:49 +0000)]
change some capitalization
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Jeff Peeler [Tue, 14 Apr 2009 15:54:16 +0000 (15:54 +0000)]
Add service maintenance message support
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
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Olle Johansson [Tue, 14 Apr 2009 14:22:39 +0000 (14:22 +0000)]
New actions should go under "New Actions", not "new events"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188284
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Olle Johansson [Tue, 14 Apr 2009 14:20:10 +0000 (14:20 +0000)]
Making sure we have references to external libraries.
Note: Update h.323 with the recent changes too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188283
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Joshua Colp [Tue, 14 Apr 2009 13:14:21 +0000 (13:14 +0000)]
Fix a bug with the change I made yesterday to outbound proxy support.
Per discussion with oej on IRC we need the actual IP address, not the
outbound proxy IP address, in the sa field. This change matches the already
existing code for all other uses of the outbound proxy setting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188247
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Tilghman Lesher [Tue, 14 Apr 2009 05:45:13 +0000 (05:45 +0000)]
As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188210
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Tilghman Lesher [Tue, 14 Apr 2009 05:27:53 +0000 (05:27 +0000)]
Application delimiter is ',', not '|'.
(closes issue #14881)
Reported by: stegro
Patches:
smsq.patch uploaded by stegro (license 752)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188206
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Tilghman Lesher [Mon, 13 Apr 2009 23:05:53 +0000 (23:05 +0000)]
Blocked revisions 188149 via svnmerge
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r188149 | tilghman | 2009-04-13 18:04:15 -0500 (Mon, 13 Apr 2009) | 4 lines
If fileconfig limit exceeds our maximum, then set the limit to the maximum.
(Closes issue #14888)
Reported by: falves11
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Mark Michelson [Mon, 13 Apr 2009 19:31:48 +0000 (19:31 +0000)]
Fix another crash related to cached realtime music on hold.
This was another off-by-one problem caused by moh_register.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188102
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Joshua Colp [Mon, 13 Apr 2009 16:28:06 +0000 (16:28 +0000)]
Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1.
Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
be sending to. This has to be done because the logic that determines what local IP address to use
in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
we are sending to.
(closes issue #12006)
Reported by: mnicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188067
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Mark Michelson [Mon, 13 Apr 2009 14:17:56 +0000 (14:17 +0000)]
Set all queue variables on both the caller and member channels.
This allows for the variables to be accessed if a member macro is run.
Thanks to Grigoriy Puzankin for bringing this up on the -dev list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188032
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Jeff Peeler [Fri, 10 Apr 2009 22:16:38 +0000 (22:16 +0000)]
Blocked revisions 187962 via svnmerge
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r187962 | jpeeler | 2009-04-10 17:16:13 -0500 (Fri, 10 Apr 2009) | 9 lines
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
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Jeff Peeler [Fri, 10 Apr 2009 20:26:46 +0000 (20:26 +0000)]
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
Patches:
issue_11966.patch uploaded by kpfleming (license 421)
Tested by: jpeeler
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Russell Bryant [Fri, 10 Apr 2009 19:27:23 +0000 (19:27 +0000)]
Blocked revisions 187865 via svnmerge
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r187865 | russell | 2009-04-10 14:26:40 -0500 (Fri, 10 Apr 2009) | 4 lines
Support "signaling" in addition to "signalling".
The sample configuration file has references to both spellings.
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Mark Michelson [Fri, 10 Apr 2009 18:56:52 +0000 (18:56 +0000)]
Indicating connected line or redirecting updates were missing a call to lock the local_pvt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187830
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Joshua Colp [Fri, 10 Apr 2009 18:14:47 +0000 (18:14 +0000)]
Change how we set the local and remote address.
The code will now only change the address and port. It will not overwrite any other values.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187773
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Joshua Colp [Fri, 10 Apr 2009 18:02:44 +0000 (18:02 +0000)]
Fix some uninitialized memory notices that appeared under valgrind.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772
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Mark Michelson [Fri, 10 Apr 2009 17:32:25 +0000 (17:32 +0000)]
Make sure tc is unlocked before calling ast_call since calling a Local
channel could result in a deadlock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187770
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Tilghman Lesher [Fri, 10 Apr 2009 17:29:34 +0000 (17:29 +0000)]
Merged revisions 187763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines
Add lastms column to the contributed table designs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187764
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Kevin P. Fleming [Fri, 10 Apr 2009 16:51:44 +0000 (16:51 +0000)]
clean up some patterns for files to remove
add embedding support for bridge and test modules
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187721
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Mark Michelson [Fri, 10 Apr 2009 16:26:48 +0000 (16:26 +0000)]
ast_strdup failures aren't really failures if the original value was NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187714
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Mark Michelson [Fri, 10 Apr 2009 16:06:22 +0000 (16:06 +0000)]
Don't let ast_channel_alloc fail if explicitly passed NULL cid_name or cid_number.
This also fixes a small memory leak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187680
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Russell Bryant [Fri, 10 Apr 2009 16:00:29 +0000 (16:00 +0000)]
Disable test modules by default.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187675
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Tilghman Lesher [Fri, 10 Apr 2009 15:59:40 +0000 (15:59 +0000)]
Ensure pvt is not NULL before dereferencing it.
(closes issue #14784)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187674
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David Vossel [Fri, 10 Apr 2009 15:49:16 +0000 (15:49 +0000)]
Even more changes concerning r187426. Revised where locks are placed yet once again. ast_call() should not be called with a channel locked. could cause deadlock issues with local channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187673
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Kevin P. Fleming [Fri, 10 Apr 2009 15:11:16 +0000 (15:11 +0000)]
revert addition of LOG_SECURITY log channel; after further discussion, a much better solution will be used
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187636
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Richard Mudgett [Fri, 10 Apr 2009 14:53:59 +0000 (14:53 +0000)]
Miscellaneous minor changes to chan_misdn.
* Miscellaneous spacing and comment changes.
* Minor code rearangements.
* Miscellaneous doxygen comments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187635
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Richard Mudgett [Fri, 10 Apr 2009 14:50:42 +0000 (14:50 +0000)]
Make chan_misdn_log() avoid generating the log message if logging is disabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187634
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Tilghman Lesher [Fri, 10 Apr 2009 03:55:27 +0000 (03:55 +0000)]
Modify headers and macros, according to Russell's suggestions on the -dev list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187599
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Mark Michelson [Thu, 9 Apr 2009 21:06:26 +0000 (21:06 +0000)]
Add a new option, mwi_from, to sip.conf.
This allows for you to change the From header for outgoing MWI
NOTIFY requests. Prior to this, the best you could do was to
set a callerid in the general section of sip.conf. The problem
was that this was used for all outbound requests, not just
MWI NOTIFY requests.
AST-201
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187560
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David Vossel [Thu, 9 Apr 2009 20:40:34 +0000 (20:40 +0000)]
More changes concerning r187426. Revised where locks are placed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187556
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Jeff Peeler [Thu, 9 Apr 2009 19:10:02 +0000 (19:10 +0000)]
Add ability for dialplan execution to continue when caller hangs up.
The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187491
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Mark Michelson [Thu, 9 Apr 2009 18:58:41 +0000 (18:58 +0000)]
Merged revisions 187484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines
Handle a SIP race condition (reinvite before an ACK) properly.
RFC 5047 explains the proper course of action to take if a
reINVITE is received before the ACK from a previous invite
transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of
the sip_pvt representing this dialog.
(closes issue #13849)
Reported by: klaus3000
Patches:
13849_v2.patch uploaded by mmichelson (license 60)
Tested by: mmichelson, klaus3000
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187488
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Tilghman Lesher [Thu, 9 Apr 2009 18:40:01 +0000 (18:40 +0000)]
Merged revisions 187428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines
Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
Add lock timeouts to avoid this potential deadlock.
(closes issue #14705)
Reported by: jamessan
Patches:
20090320__bug14705.diff.txt uploaded by tilghman (license 14)
Tested by: jamessan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187483
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David Vossel [Thu, 9 Apr 2009 17:39:10 +0000 (17:39 +0000)]
Fixes deadlock caused by calling get_cid_name with chan locked.
get_cid_name should not be called with a channel lock. get_cid_name calls ast_get_hint which eventually calls pbx_find_extension. pbx_find_extension starts and stops autoservice which should not be done with a channel lock, so get_cid_name should not be called with one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187426
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Mark Michelson [Thu, 9 Apr 2009 17:34:39 +0000 (17:34 +0000)]
Use safe macro practices even though they really aren't necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187424
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Mark Michelson [Thu, 9 Apr 2009 17:30:39 +0000 (17:30 +0000)]
Fix a crash in res_musiconhold when using cached realtime moh.
The moh_register function links an mohclass and then immediately
unrefs the class since the container now has a reference. The problem
with using realtime music on hold is that the class is allocated,
registered, and started in one fell swoop. The refcounting logic
resulted in the count being off by one. The same problem did not
happen when using a static config because the allocation and registration
of an mohclass is a separate operation from starting moh. This also did
not affect non-cached realtime moh because the classes are not registered
at all.
I also have modified res_musiconhold to use the _t_ variants of the ao2_
functions so that more info can be gleaned when attempting to trace the
refcounts. I found this to be incredibly helpful for debugging this issue
and there's no good reason to remove it.
(closes issue #14661)
Reported by: sum
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187421
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Tilghman Lesher [Thu, 9 Apr 2009 17:20:49 +0000 (17:20 +0000)]
Allow '/' in username portion of register; this is a regression.
(closes issue #14668)
Reported by: Netview
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187381
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Tilghman Lesher [Thu, 9 Apr 2009 16:39:43 +0000 (16:39 +0000)]
Merged revisions 187362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines
Permit zero-length text messages in SIP.
(Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal")
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187363
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Joshua Colp [Thu, 9 Apr 2009 16:27:53 +0000 (16:27 +0000)]
Do not try to send the format read/format write/make compatible options over IAX2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187361
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Joshua Colp [Thu, 9 Apr 2009 16:19:35 +0000 (16:19 +0000)]
Add support for allowing the channel driver to handle transcoding.
This was accomplished using a set of options and the setoption channel callback.
The core calls into the channel driver using these options and the channel driver
either returns success or failure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187360
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Tilghman Lesher [Thu, 9 Apr 2009 04:59:05 +0000 (04:59 +0000)]
Merged revisions 187300-187301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625)
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r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
Oops, missed this file in the last commit.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187302
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Kevin P. Fleming [Thu, 9 Apr 2009 02:44:27 +0000 (02:44 +0000)]
add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187269
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Jeff Peeler [Wed, 8 Apr 2009 21:00:39 +0000 (21:00 +0000)]
Add timer for features so that backup bridge config can go away
The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
(closes issue #14503)
Reported by: KNK
Tested by: jpeeler
Review: http://reviewboard.digium.com/r/179/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187211
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Tilghman Lesher [Wed, 8 Apr 2009 20:39:55 +0000 (20:39 +0000)]
Recorded merge of revisions 187209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009) | 4 lines
Backport resolution for file descriptor leak in 1.6.0 to 1.4.
This fixes short reads in http manager sessions, such as those done by the
ast-gui branch. (Fixes AST-198)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187210
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Russell Bryant [Wed, 8 Apr 2009 19:59:21 +0000 (19:59 +0000)]
Add documentation for reviewboard usage and guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187179
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Mark Michelson [Wed, 8 Apr 2009 19:18:10 +0000 (19:18 +0000)]
Blocked revisions 187135 via svnmerge
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r187135 | mmichelson | 2009-04-08 14:16:49 -0500 (Wed, 08 Apr 2009) | 8 lines
Fix a crash due to too few arguments to RetryDial.
(closes issue #14852)
Reported by: junky
Patches:
retry_fix.diff uploaded by junky (license 177)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187138
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Joshua Colp [Wed, 8 Apr 2009 18:12:28 +0000 (18:12 +0000)]
Fix a bug where we would native bridge when we did not want to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187108
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Russell Bryant [Wed, 8 Apr 2009 17:51:35 +0000 (17:51 +0000)]
Remove duplicate prototype for temp_peer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187105
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Tilghman Lesher [Wed, 8 Apr 2009 17:08:43 +0000 (17:08 +0000)]
If the first column is empty, output a delimiter anyway.
(closes issue #14848)
Reported by: john8675309
Patches:
20090408__bug14848.diff.txt uploaded by tilghman (license 14)
Tested by: john8675309
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187050
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Mark Michelson [Wed, 8 Apr 2009 16:52:20 +0000 (16:52 +0000)]
Merged revisions 187045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187046
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Joshua Colp [Wed, 8 Apr 2009 16:27:36 +0000 (16:27 +0000)]
Turn a warning message into a debug message and do not treat two situations as errors when they are not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187036
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Mark Michelson [Wed, 8 Apr 2009 15:27:41 +0000 (15:27 +0000)]
Merged revisions 186984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines
Make a couple of changes with regards to a new message printed in ast_read().
"ast_read() called with no recorded file descriptor" is a new message added
after a bug was discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this error message
to be displayed. This commit does two things to help to make this message appear
less.
First, the message has been downgraded to a debug level message if dev mode is
not enabled. The message means a lot more to developers than it does to end users,
and so developers should take an effort to be sure to call ast_read only when
a channel is ready to be read from. However, since this doesn't actually cause an
error in operation and is not something a user can easily fix, we should not spam
their console with these messages.
Second, the message has been moved to after the check for any pending masquerades.
ast_read() being called with no recorded file descriptor should not interfere with
a masquerade taking place.
This could be seen as a simple way of resolving issue #14723. However, I still want
to try to clear out the existing ways of triggering this message, since I feel that
would be a better resolution for the issue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186985
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Russell Bryant [Wed, 8 Apr 2009 13:38:27 +0000 (13:38 +0000)]
Add some additional notes on release numbering.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186957
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Russell Bryant [Wed, 8 Apr 2009 13:24:48 +0000 (13:24 +0000)]
Start splitting up miscellaneous doxygen documentation into separate files.
doxyref.h was created to hold miscellaneous documentation that was not specific
to a part of the code. This file has grown quite a bit so I decided to start
splitting parts of it out into new files. Now, you can drop a new file into
include/asterisk/doxygen/ and it will be processed by doxygen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186953
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Russell Bryant [Wed, 8 Apr 2009 12:35:57 +0000 (12:35 +0000)]
Update some comments and resolve potential memory corruption in chan_sip.
While browsing chan_sip the other day, I noticed this dangerous code in
dialog_needdestroy(). This function is an ao2_callback. It is absolutely
_not_ okay to unlock the container from within this function. It's also not
clear why it was useful. Given that it could cause memory corruption, I have
removed it.
There was also a TODO comment left describing a potential implementation of
an improvement to the needdestroy handling. I'm not convinced that what was
described is the best choice here, so I have briefly described the way that
this function is used today that could be improved.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186928
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Tilghman Lesher [Wed, 8 Apr 2009 05:06:22 +0000 (05:06 +0000)]
Add lastms to the require API call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186899
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Mark Michelson [Wed, 8 Apr 2009 00:09:28 +0000 (00:09 +0000)]
Merged revisions 186841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines
Fix a few typos of the word "frequency."
(closes issue #14842)
Reported by: jvandal
Patches:
frequency-typo.diff uploaded by jvandal (license 413)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186842
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Mark Michelson [Wed, 8 Apr 2009 00:01:49 +0000 (00:01 +0000)]
Fix bad merge from fix for issue 13867.
(closes issue #14686)
Reported by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186837
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Mark Michelson [Tue, 7 Apr 2009 23:50:56 +0000 (23:50 +0000)]
Merged revisions 186832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines
Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY.
Without this flag set, warning sounds will not be properly played to either party
of the bridge.
(closes issue #14845)
Reported by: adomjan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186833
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Tilghman Lesher [Tue, 7 Apr 2009 22:23:46 +0000 (22:23 +0000)]
Merged revisions 186775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines
Fix Macro documentation to match current (and intended) behavior.
(See -dev mailing list)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186799
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Mark Michelson [Tue, 7 Apr 2009 20:46:18 +0000 (20:46 +0000)]
Merged revisions 186719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines
Ensure that \r\n is printed after the ActionID in an OriginateResponse.
(closes issue #14847)
Reported by: kobaz
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186720
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Joshua Colp [Mon, 6 Apr 2009 23:11:13 +0000 (23:11 +0000)]
Fix a log message getting output when it should not have been.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186687
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Joshua Colp [Mon, 6 Apr 2009 17:03:07 +0000 (17:03 +0000)]
Fix problem when authenticating a non-RTP dialog.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186653
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Joshua Colp [Mon, 6 Apr 2009 16:15:30 +0000 (16:15 +0000)]
Add support for changing the outbound codec on a SIP call using
a dialplan variable.
This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.
(closes issue #13243)
Reported by: samdell3
Patches:
13243.diff uploaded by file (license 11)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624
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Mark Michelson [Mon, 6 Apr 2009 16:06:25 +0000 (16:06 +0000)]
Silly svn. These files didn't get merged over in the merge of the issue8824 branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186620
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Mark Michelson [Mon, 6 Apr 2009 13:57:39 +0000 (13:57 +0000)]
Blocked revisions 186565 via svnmerge
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r186565 | mmichelson | 2009-04-06 08:54:41 -0500 (Mon, 06 Apr 2009) | 3 lines
Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used.
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Joshua Colp [Mon, 6 Apr 2009 13:23:12 +0000 (13:23 +0000)]
Pass the correct value to sizeof when copying address information.
(issue #14827)
Reported by: pj
Patches:
14827.diff uploaded by file (license 11)
Tested by: pj
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Richard Mudgett [Sat, 4 Apr 2009 00:13:50 +0000 (00:13 +0000)]
Remove merged branch properties accidentally merged to trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186537
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Mark Michelson [Fri, 3 Apr 2009 22:41:46 +0000 (22:41 +0000)]
This commit introduces COLP/CONP and Redirecting party information into Asterisk.
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
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Kevin P. Fleming [Fri, 3 Apr 2009 20:20:01 +0000 (20:20 +0000)]
Merged revisions 186458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines
Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested
Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later).
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Tilghman Lesher [Fri, 3 Apr 2009 19:59:55 +0000 (19:59 +0000)]
Merged revisions 186445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines
Found a conflict in the last commit, due to multiple targets
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Tilghman Lesher [Fri, 3 Apr 2009 19:30:34 +0000 (19:30 +0000)]
Merged revisions 186415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines
Distinguish in a sent email between simple sends and forwards.
(closes issue #11678)
Reported by: jamessan
Patches:
20090330__bug11678.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, lmadsen
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Joshua Colp [Fri, 3 Apr 2009 16:47:27 +0000 (16:47 +0000)]
Add better support for relaying success or failure of the ast_transfer() API call.
This API call now waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to truly convey whether
the transfer worked or not. In the case of the Transfer() dialplan application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.
(closes issue #12713)
Reported by: davidw
Tested by: file
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