Commit Graph

21149 Commits

Author SHA1 Message Date
Alexandr Anikin bd32cb4ba5 lc not found - it's warning, not error,
change malloc to ast_calloc again


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-13 10:50:22 +00:00
Alexandr Anikin eaf73d6588 change malloc to ast_calloc calls to prevent crash of asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-12 23:25:58 +00:00
Jason Parker 96cbd4ffcd Merged revisions 307536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-10 22:43:51 +00:00
Mark Michelson 4cba13eb60 Merged revisions 307467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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
2011-02-10 17:45:24 +00:00
David Vossel 08460fc094 Fixes bug in chan_sip where nativeformats are not set correctly.
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
2011-02-10 17:12:10 +00:00
Alexandr Anikin 707cf78c5a Corrections for properly work with H.323v2 (older) endpoints and other
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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-10 13:29:19 +00:00
Jeff Peeler 8f7982f280 Add new manager action MeetmeListRooms.
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
2011-02-09 22:48:02 +00:00
Andrew Latham 0703a9a321 Disable color during running test
(closes issue #18776)
Reported by: alecdavis
Patches:
     ast_deb_init.diff uploaded by lathama (license 1028)
Tested by: andrel, lathama


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09 21:46:24 +00:00
Jeff Peeler 10362292ef Merged revisions 307273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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
2011-02-09 21:08:22 +00:00
Jeff Peeler e2df246636 Allow parkedmusicclass to be settable for non-default parking lots.
(closes issue #17946)
Reported by: bluecrow76
Patches:
      asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09 20:11:11 +00:00
Jeff Peeler 6b0fa46103 Merged revisions 307228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09 19:53:28 +00:00
Tzafrir Cohen 1540401a4a clarify warning when no loadable module support
Clarify warning message when LOADABLE_MODULES is disabled but we still
try to load a module.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09 19:17:01 +00:00
Tilghman Lesher fc4df44bd8 Merged revisions 307142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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.)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09 05:53:29 +00:00
Jason Parker f01e9568d2 Merged revisions 307092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 21:24:57 +00:00
Mark Michelson 0074165356 Merged revisions 307065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 21:18:26 +00:00
Terry Wilson 4f57a3bb7c Merged revisions 306979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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
2011-02-08 20:42:44 +00:00
Andrew Latham a350924700 Documentation Updates
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



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 20:31:13 +00:00
Jeff Peeler a46bfe67bd Merged revisions 306967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 19:42:03 +00:00
Jeff Peeler e2cdaf47bb Merged revisions 306962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 19:26:05 +00:00
Jeff Peeler 9264ab00f5 Merged revisions 306866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 16:22:07 +00:00
Andrew Latham c4271440ab Documentation Updates.
Start updates to the man pages.

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 02:05:03 +00:00
Richard Mudgett 8b584000a9 Define the MCID acronym in chan_dahdi.conf.sample.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 00:43:34 +00:00
Richard Mudgett 209a39f4b0 Use correct conditional for MCID send.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 00:26:01 +00:00
Richard Mudgett 49feb747ba Pass a MCID request to the bridged channel.
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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07 23:33:44 +00:00
Terry Wilson 1277a80a5b Merged revisions 306674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07 22:46:07 +00:00
Terry Wilson a974d1a4ce Merged revisions 306619 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07 22:31:25 +00:00
Mark Michelson f4ea670a6a Merged revisions 306575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07 17:55:38 +00:00
David Vossel 2db3c9e058 Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07 16:33:43 +00:00
Alexandr Anikin 7f86bd2f16 fix trivial issue after dvossel patch, initial zero fill user and peer
structure before cap structure allocated.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-05 22:16:07 +00:00
Richard Mudgett 484f9bec0a Ignore voice frames in chan_dahdi native bridging. Hardware is handling them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-05 02:55:50 +00:00
Jeff Peeler 3d667d7c0f Send manager event for blackfilter only if it DOES NOT match.
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
2011-02-04 22:37:11 +00:00
Richard Mudgett a8aeb04a9f Add ISDN display ie text handling options to chan_dahdi.conf.
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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 20:30:48 +00:00
Jason Parker 0beeb00ef3 Merged revisions 306356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 19:24:54 +00:00
Richard Mudgett 9fc0d9be66 Fix compiler warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 19:09:00 +00:00
Richard Mudgett 4d8feab7fa Merged revisions 306324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 18:57:39 +00:00
Paul Belanger ed3a4856aa Revert changes to extconf.c
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
2011-02-04 18:16:16 +00:00
Paul Belanger 3556e4c2d4 Replace ast_log(LOG_DEBUG, ...) with ast_debug()
(closes issue #18556)
Reported by: kkm

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 16:55:39 +00:00
David Vossel 9f65acf33e Fix compile error in codec ilbc translator.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04 16:42:15 +00:00
Jeff Peeler 285d953fdf Merged revisions 306215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 23:50:08 +00:00
Terry Wilson 36da6b6286 Merged revisions 306127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines
  
  Merged revisions 306126 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines
    
    Merged revisions 306119 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines
      
      Set hangup cause in local_hangup
      
      When a call involves a local channel (like SIP -> Local -> SIP), the hangup
      cause was not being set. This resulted in SIP channels sometimes getting a
      503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+
      this also can cause issues with CCSS that involve a local channel. This patch
      sets the hangupcause for one side of the local channel to the other in
      local_hangup for outbound calls.
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 21:13:11 +00:00
Jeff Peeler fed10ed35d Merged revisions 306124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306124 | jpeeler | 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) | 17 lines
  
  Merged revisions 306123 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines
    
    Set exception on channel in parking thread when POLLPRI event detected.
    
    This is done just to make the code be equivalent to the old select code. As
    noted in 303106 the same issue was already fixed in this branch, but the
    exception was not set on the channel in the case of POLLPRI. The reason that
    this did not cause a problem here is because in 122923 the check in __ast_read
    to check the exception flag was removed.
    
    (related to #18637)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 20:51:09 +00:00
Jason Parker 7f76b3d573 Modify alignment of 'core show codecs', since the ID is no longer a huge int.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 18:37:06 +00:00
David Vossel 63f5a80a3b Fixes output of "core show codecs" to display image types correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 18:12:57 +00:00
David Vossel c26c190711 Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:22:10 +00:00
Andrew Latham 652fb64a01 res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
(issue #18713)
Reported by: lathama
Patches:
     snom_dir.diff uploaded by lathama (license 1028)
Tested by: lathama


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:13:40 +00:00
Richard Mudgett f71322f239 Merged revisions 305923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines
  
  Merged revisions 305889 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
    
    Merged revisions 305888 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines
    
      Minor AST_FRAME_TEXT related issues.
    
      * Include the null terminator in the buffer length.  When the frame is
      queued it is copied.  If the null terminator is not part of the frame
      buffer length, the receiver could see garbage appended onto it.
    
      * Add channel lock protection with ast_sendtext().
    
      * Fixed AMI SendText action ast_sendtext() return value check.
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 00:29:46 +00:00
Tilghman Lesher 2740326200 Merged revisions 305844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) | 5 lines
  
  Eliminate a file descriptor leak when using the FILE() dialplan function.
  
  (closes issue #18731)
  Reported by: marioabajo
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02 20:06:33 +00:00
Andrew Latham 93bade5639 Replacing doc/* and asterisk.pdf with wiki links
Adding links to http(s)://wiki.asterisk.org



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02 19:30:49 +00:00
Andrew Latham 9f1a17f137 Replacing doc/* with wiki links
Adding links to http(s)://wiki.asterisk.org



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02 18:59:29 +00:00
Andrew Latham 175dd0ebf6 Replace link to old doc with new wiki page.
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02 15:25:12 +00:00