https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines
  
  Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
  
  A bug in AEL did not distinguish between the "s" extension generated by
  AEL and an "s" extension that was required to exist by the chan_dahdi
  (or another channel) that was not supplied with a starting extension.
  Therefore, AEL made incorrect assumptions about what commands were
  permissable in the context.  This was fixed by making AEL generate a
  different extension name.  However, Dial and Queue make additional
  assumptions about the name of the default gosub extension.  Therefore,
  they needed to be brought into line with a "macro" rendered by AEL (as
  a gosub), without breaking traditional dialplans written without the
  aid of AEL.
  
  Related to (issue #18480)
   Reported by: nivek
  
  (closes issue #18729)
   Reported by: kkm
   Patches: 
         20110209__issue18729.diff.txt uploaded by tilghman (license 14)
         018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
   Tested by: kkm
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

change malloc to ast_calloc again


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  r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines
  
  Merged revisions 307535 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines
    
    Merged revisions 307534 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines
      
      Remove color when executing commands via a remote console.
      
      Essentially this makes '-x' imply '-n' on rasterisk.  This was done in a
      different and incomplete way previously, which I'm reverting here.
      
      (issue #18776)
      Reported by: alecdavis
    ........
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  r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines
  
  Fix a gaffe in the CCSS sample configuration.
  
  Discovered by Philippe Lindheimer and pointed out on #asterisk-dev
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307468 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The nativeformats field was being overwritten when it should have been
appended too.  This caused some format capabilities to be lost briefly and
some log warnings to be output.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3

small fixes.

Interpret remote side H.225 version.

Corrections for H.323v2 endpoints: 
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.

Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future), 
don't force H.245tunneling if FastStart is active, don't send Alerting 
singal more than once per call.

(closes issue #18542)
Reported by: vmikhelson
Patches: 
      issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson


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From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.

(closes issue #17905)
Reported by: rcasas
Patches: 
      app_meetme.c.patch uploaded by rcasas (license 641)

Review: https://reviewboard.asterisk.org/r/874/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #18776)
Reported by: alecdavis
Patches:
     ast_deb_init.diff uploaded by lathama (license 1028)
Tested by: andrel, lathama


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  r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) | 8 lines
  
  Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback.
  
  (closes issue #18758)
  Reported by: rgagnon
  Patches: 
        branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
        trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17946)
Reported by: bluecrow76
Patches:
      asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff


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  r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines
  
  Merged revisions 307227 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines
    
    Make sure to set parking dial context for non-default parking lots.
    
    Since parking_con_dial isn't settable, set all parking lots to "park-dial".
    
    (closes issue #17946)
    Reported by: bluecrow76
    Patches:
          asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270)
          modified by me
  ........
................


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Clarify warning message when LOADABLE_MODULES is disabled but we still
try to load a module.

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  r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) | 3 lines
  
  Initialize tracking variable in structure properly.  Fixes a memory leak.
  (Reported by The_Boy_Wonder on IRC, fixed by me.)
........


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  r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines
  
  Fix issue with verbose messages not showing on remote console.
  
  This code was reworked recently, and since the logchannel list hadn't been
  created yet at this point, and it was a verbose message, it was being dropped
  on the floor.  Now it'll continue on to where it should be handled.
  
  (closes issue #18580)
  Reported by: pabelanger
........


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  r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines
  
  Add a couple of useful channel variables for the CC recall macro.
  
  CC_EXTEN and CC_CONTEXT will allow you to determine the channel
  and context that will be called when the recall occurs.
........


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  r306979 | twilson | 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) | 16 lines
  
  Merged revisions 306973 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines
    
    Merged revisions 306972 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines
      
      Fix comparison for REFER Replaces tags with pedantic=yes
    ........
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Note default polling setting in voicemail.conf
Add missing config to asterisk.conf
Update manpage

(issue #16505)
Reported by: tzafrir
Patches:
     asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir



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  r306967 | jpeeler | 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) | 16 lines
  
  Merged revisions 306966 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines
    
    Merged revisions 306965 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line
      
      fix this line again
    ........
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  r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines
  
  Merged revisions 306961 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines
    
    Merged revisions 306960 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines
      
      Backup file storing message duration is not used with IMAP_STORAGE, remove code.
      
      The message duration is stored in the body of the email when using IMAP_STORAGE,
      so nothing needs to happen with the backup file.
      
      (closes issue #18718)
      Reported by: kerframil
    ........
  ................
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  r306866 | jpeeler | 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) | 16 lines
  
  Merged revisions 306865 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines
    
    Merged revisions 306864 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line
      
      make this safer and fully correct, pointed out by Steve Davis
    ........
  ................
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Start updates to the man pages.

(issue #16505)
Reported by: tzafrir
Tested by: lathama


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Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.

The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.

JIRA SWP-2845
JIRA ABE-2736


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  r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines
  
  Merged revisions 306673 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines
    
    Merged revisions 306672 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines
      
      Don't try to pickup a call in the middle of a masquerade
      
      If A calls B which doesn't answer and C & D both try to do a call pickup, it is
      possible for ast_pickup_call to answer the call, then fail to masquerade one of
      the calls because the other one is already in the process of masquerading. This
      patch checks to see if the channel is in the process of masquerading before
      call before selecting it for a pickup.
      
      Review: https://reviewboard.asterisk.org/r/1094/
    ........
  ................
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  r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines
  
  Merged revisions 306618 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines
    
    Merged revisions 306617 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines
      
      Don't allow a REFER w/replaces to replace its own dialog
      
      Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces
      header that matches the dialog of the REFER. This would be a situation like A
      calls B, A calls C, A transfers B to A, which is just silly. This patch makes
      the transfer fail instead of making Asterisk freak out and forget to hang other
      channels up.
      
      Review: https://reviewboard.asterisk.org/r/1093/
    ........
  ................
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  r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines
  
  Rearrange a bit of code in the generic CC recall operation.
  
  By waiting to call the callback macro after the CC_INTERFACES,
  extension, priority, and context have been set, this information
  can be accessed more easily within the callback macro.
  
  Reported by Philippe Lindheimer.
........


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structure before cap structure allocated.


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The logic got reversed, oops. Works properly now when multiple blackfilters are
present.

(closes issue #18283)
Reported by: telecos82
Patches: 
      ast_managereventfilter.patch uploaded by telecos82 (license 687)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The display ie handling can be controlled independently in the send and
receive directions with the following options:

* Block display text data.

* Use display text in SETUP/CONNECT messages for name.

* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).

* Pass arbitrary display text during a call.  Sent in INFORMATION
messages.  Received from any message that the display text was not used as
a name.

If the display options are not set then the options default to legacy
behavior.

The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.

To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.

JIRA SWP-2688
JIRA ABE-2693


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  r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines
  
  Merged revisions 306346 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines
    
    Don't fallthrough to 'unknown' in the 'ringing' case.
    
    This could cause improper exits from the queue.
    
    (closes issue #18499)
    Reported by: zaltar
    Patches: 
          app_queue.patch uploaded by zaltar (license 1148)
  ........
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  r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines
  
  Don't send redirecting updates to the caller if the dialplan forked the call.
  
  Each fork in the dial could be redirected and confuse the caller.  For
  ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN
  redirects calls in sequence not in parallel.
  
  * Also fixed a formatting inconsistency in app_dial.c and make a warning
  message more useful about what frame type could not be written.
........


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It seems extconf.c already defines some local ast_debug() functions.  Theses
should be removed and replaced with logger.h.  A patch will be added to
reviewboard shortly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #18556)
Reported by: kkm

Review: https://reviewboard.asterisk.org/r/1071/


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........
  r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines
  
  Fix SIP deadlock involving state changes.
  
  Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
  has caused locking problems. Both of these functions lock the channel when
  the channel argument is passed in!
  
  In this case, the suspected problem (the backtrace makes it impossible to tell)
  was the private being locked in sip_set_rtp_peer and then:
  transmit_reinvite_with_sdp
   try_suggested_sip_codec
     pbx_builtin_getvar_helper
  (Traced to verify that the fix was only required in 1.8 and later.)
  
  (closes issue #18491)
  Reported by: cmaj
  Patches: 
        chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830)
  Tested by: cmaj
........


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