Commit Graph

412 Commits

Author SHA1 Message Date
Tilghman Lesher
51889e1ff7 Merged revisions 244393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r244393 | tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18 lines
  
  Properly respect GOSUB_RESULT as to what to do with the master channel.
  
  Previously, we would parse GOSUB_RESULT, but not actually do anything with it.
  
  (closes issue #16686)
   Reported by: bklang
   Patches: 
         app_dial-respect-gosub_result.patch uploaded by bklang (license 919)
         (with modifications)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@244394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-02 20:35:35 +00:00
Matthew Nicholson
b464edd1c7 Merged revisions 227829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines
  
  Merged revisions 227827 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines
    
    This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements.
    
    (closes issue #16005)
    Reported by: falves11
    Patches:
          dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96)
    Tested by: mnicholson, falves11
    
    Review: https://reviewboard.asterisk.org/r/407/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 21:15:46 +00:00
Joshua Colp
c968818ffb Merged revisions 226890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines
  
  Merged revisions 226889 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
    
    Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
    while the called party had not yet answered.
    
    This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
    file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
    
    (closes issue #14674)
    Reported by: ulogic
    Patches:
          bug14674.patch uploaded by jpeeler (license 325)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@226892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-02 18:11:19 +00:00
Joshua Colp
eb20b22f65 Merged revisions 224567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines
  
  Merged revisions 224565 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines
    
    Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.
    
    (closes issue #14763)
    Reported by: cupotka
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19 19:51:12 +00:00
Jeff Peeler
445d7f7e50 Merged revisions 223832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines
  
  Merged revisions 223804 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines
    
    Ensure ringing continues for branched calls after progress is received
    
    While waiting for an answer, don't send progress for branched calls
    for which ringing was sent.
    
    (closes issue #15028)
    Reported by: fnordian
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-12 23:55:07 +00:00
Mark Michelson
0fd3ac4508 Merged revisions 223215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines
  
  Recorded merge of revisions 223213 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines
    
    Fix potential memory leak in app_dial.c
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-09 18:25:19 +00:00
Russell Bryant
3ca57ec466 Merged revisions 208593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines
  
  Merged revisions 208592 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines
    
    Do not log an ERROR if autoservice_stop() returns -1.
    
    This does not indicate an error.  A return of -1 just means that the channel
    has been hung up.
    
    (reported in #asterisk-dev)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-24 18:52:52 +00:00
Sean Bright
6a3d973648 Merged revisions 198285 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines
  
  Merged revisions 198251 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines
    
    Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
    
    (closes issue #15056)
    Reported by: p_lindheimer
    Patches:
          05292009_bug15056.diff uploaded by seanbright (license 71)
    Tested by: p_lindheimer
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-30 03:28:05 +00:00
Eliel C. Sardanons
2f92da1e7e Merged revisions 195162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
  
  Warn about the use of the application WaitExten() within a Macro().
  
  Update applications documentation to warn the user about the use of the
  WaitExten() application within a Macro(). Recommend the use of Read()
  instead.
  
  (closes issue #14444)
  Reported by: ewieling
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 15:13:34 +00:00
Terry Wilson
16b504cf72 Merged revisions 189516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r189516 | twilson | 2009-04-20 16:29:29 -0500 (Mon, 20 Apr 2009) | 9 lines
  
  Merged revisions 189465 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-20 21:38:11 +00:00
Terry Wilson
c4cbbb01a8 Merged revisions 189495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r189495 | twilson | 2009-04-20 16:24:34 -0500 (Mon, 20 Apr 2009) | 9 lines
  
  Merged revisions 189463 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-20 21:36:49 +00:00
David Vossel
cad3e27d20 Merged revisions 183436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines
  
  Merged revisions 183386 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
    
    Cleaning up a few things in detect disconnect patch
    
    Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory.  Cleaned up /param tags in features.h.  No longer send dynamic features in ast_feature_detect. 
    
    issue #11583
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 20:33:19 +00:00
David Vossel
88af25b2bd Merged revisions 183172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines
  
  Merged revisions 183126 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
    
    Allow disconnect feature before a call is bridged
    
    feature.conf has a disconnect option.  By default this option is set to '*', but it could be anything.  If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else.  This is because features are unavailable until bridging takes place.  The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different.  This patch allows features to be detected from outside of the bridge, but not operated on.  In this case, the disconnect feature can be detected before briding and handled outside of features.c.
    
    (closes issue #11583)
    Reported by: sobomax
    Patches:
    	patch-apps__app_dial.c uploaded by sobomax (license 359)
    	11583.latest-patch uploaded by murf (license 17)
    	detect_disconnect.diff uploaded by dvossel (license 671)
    Tested by: sobomax, dvossel
    Review: http://reviewboard.digium.com/r/195/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 17:08:23 +00:00
Joshua Colp
4340fb2aa6 Merged revisions 181612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines
  
  Fix crash when sleep and retries argument was not given to RetryDial application.
  
  (closes issue #14647)
  Reported by: sherpya
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-12 13:28:39 +00:00
Joshua Colp
4b09db51ab Merged revisions 180120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
  
  Remove duplicate 'k' and 'K' Dial options.
  
  (closes issue #14601)
  Reported by: alecdavis
  Patches:
        app_dial.optionk.diff.txt uploaded by alecdavis (license 585)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@180122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04 14:41:03 +00:00
Mark Michelson
20655a3a05 Merged revisions 174945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines
  
  Fix 'd' option for app_dial and add new option to Answer application
  
  The 'd' option would not work for channel types which use RTP to transport
  DTMF digits. The only way to allow for this to work was to answer the channel
  if we saw that this option was enabled.
  
  I realized that this may cause issues with CDRs, specifically with giving false
  dispositions and answer times. I therefore modified ast_answer to take another
  parameter which would tell if the CDR should be marked answered.
  
  I also extended this to the Answer application so that the channel may be answered
  but not CDRified if desired.
  
  I also modified app_dictate and app_waitforsilence to only answer the channel if it
  is not already up, to help not allow for faulty CDR answer times.
  
  All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
  the changes except for the change to the Answer application will go in since we do
  not introduce new features into stable branches
  
  (closes issue #14164)
  Reported by: DennisD
  Patches:
        14164.patch uploaded by putnopvut (license 60)
  Tested by: putnopvut
  
  Review: http://reviewboard.digium.com/r/145
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@174947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-11 22:55:16 +00:00
Terry Wilson
ef566503d3 Merged revisions 172580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines
  
  Merged revisions 172517 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
    
    Fix feature inheritance with builtin features
    
    When using builtin features like parking and transfers, the AST_FEATURE_* flags
    would not be set correctly for all instances when either performing a builtin
    attended transfer, or parking a call and getting the timeout callback.  Also,
    there was no way on a per-call basis to specify what features someone should
    have on picking up a parked call (since that doesn't involve the Dial() command).
    There was a global option for setting whether or not all users who pickup a
    parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
    AUTOMON, or PARKCALL.
    
    This patch:
    1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
    dialplan or with setvar in channels that support it.  This variable can be set
    to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
    equivalent dial options), to set what features should be activated on this
    channel.  The patch moves the setting of the features datastores into the
    bridging code instead of app_dial to help facilitate this.
    
    2) adds global options parkedcallparking, parkedcallhangup, and
    parkedcallrecording to be similar to the parkedcalltransfers option for
    globally setting features.
    
    3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
    extension since tracking everything through multiple masquerades, etc. is
    difficult and error-prone
    
    4) attempts to fix all cases of return calls from parking and completed builtin
    transfers not having the correct permissions
    (closes issue #14274)
    Reported by: aragon
    Patches: 
          fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
    Tested by: aragon, otherwiseguy
    
    Review http://reviewboard.digium.com/r/138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@172636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-31 00:05:17 +00:00
Joshua Colp
692b93004e Merged revisions 170569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170568 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.
    (closes issue #14310)
    Reported by: RadicAlish
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 19:10:43 +00:00
Terry Wilson
a671c82ec2 Merged revisions 167935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines
  
  Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set
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2009-01-09 00:45:37 +00:00
Mark Michelson
45ac5e6838 Merged revisions 166861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines

Update app_queue to deal with the removal of AST_PBX_KEEPALIVE

When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.

I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29 18:16:10 +00:00
Steve Murphy
a7aeaf341b Merged revisions 166665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

This merged from trunk with no conflicts. I tested
mostly the 'tired' cases, and for the most part
ignored the tests for reconnecting and dialing in
to fetch a parked call, after the first case.

................
  r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines
  
  Merged revisions 166093 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  In order to merge this 1.4 patch into trunk,
  I had to resolve some conflicts and wait for
  Russell to make some changes to res_agi.
  I re-ran all the tests; 39 calls in all, and
  made fairly careful notes and comparisons: I
  don't want this to blow up some aspect of 
  asterisk; I completely removed the KEEPALIVE
  from the pbx.h decls. The first 3 scenarios
  involving feature park; feature xfer to 700;
  hookflash park to Park() app call all behave
  the same, don't appear to leave hung channels,
  and no crashes.
  
  ........
    r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
    
    This merges the masqpark branch into 1.4
    
    These changes eliminate the need for (and use of)
    the KEEPALIVE return code in res_features.c;
    There are other places that use this result code
    for similar purposes at a higher level, these appear
    to be left alone in 1.4, but attacked in trunk.
    
    The reason these changes are being made in 1.4, is
    that parking ends a channel's life, in some situations,
    and the code in the bridge (and some other places),
    was not checking the result code properly, and dereferencing
    the channel pointer, which could lead to memory corruption
    and crashes.
    
    Calling the masq_park function eliminates this danger 
    in higher levels.
    
    A series of previous commits have replaced some parking calls
    with masq_park, but this patch puts them ALL to rest,
    (except one, purposely left alone because a masquerade
    is done anyway), and gets rid of the code that tests
    the KEEPALIVE result, and the NOHANGUP_PEER result codes.
    
    While bug 13820 inspired this work, this patch does
    not solve all the problems mentioned there.
    
    I have tested this patch (again) to make sure I have
    not introduced regressions. 
    
    Crashes that occurred when a parked party hung up
    while the parking party was listening to the numbers
    of the parking stall being assigned, is eliminated.
    
    These are the cases where parking code may be activated:
    
    1. Feature one touch (eg. *3)
    2. Feature blind xfer to parking lot (eg ##700)
    3. Run Park() app from dialplan (eg sip xfer to 700)
       (eg. dahdi hookflash xfer to 700)
    4. Run Park via manager.
    
    The interesting testing cases for parking are:
    I. A calls B, A parks B
        a. B hangs up while A is getting the numbers announced.
        b. B hangs up after A gets the announcement, but 
           before the parking time expires
        c. B waits, time expires, A is redialed,
           A answers, B and A are connected, after
           which, B hangs up.
        d. C picks up B while still in parking lot.
    
    II. A calls B, B parks A
        a. A hangs up while B is getting the numbers announced.
        b. A hangs up after B gets the announcement, but 
           before the parking time expires
        c. A waits, time expires, B is redialed,
           B answers, A and B are connected, after
           which, A hangs up.
        d. C picks up A while still in parking lot.
    
    Testing this throroughly involves acting all the permutations
    of I and II, in situations 1,2,3, and 4.
    
    Since I added a few more changes (ALL references to KEEPALIVE in the bridge
    code eliimated (I missed one earlier), I retested
    most of the above cases, and no crashes.
    
    H-extension weirdness.
    
    Current h-extension execution is not completely
    correct for several of the cases.
    
    For the case where A calls B, and A parks B, the
    'h' exten is run on A's channel as soon as the park
    is accomplished. This is expected behavior.
    
    But when A calls B, and B parks A, this will be
    current behavior:
    
    After B parks A, B is hung up by the system, and
    the 'h' (hangup) exten gets run, but the channel
    mentioned will be a derivative of A's...
    
    Thus, if A is DAHDI/1, and B is DAHDI/2,
    the h-extension will be run on channel
    Parked/DAHDI/1-1<ZOMBIE>, and the 
    start/answer/end info will be those 
    relating to Channel A.
    
    And, in the case where A is reconnected to
    B after the park time expires, when both parties
    hang up after the joyful reunion, no h-exten
    will be run at all.
    
    In the case where C picks up A from the 
    parking lot, when either A or C hang up,
    the h-exten will be run for the C channel.
    
    CDR's are a separate issue, and not addressed
    here.
    
    As to WHY this strange behavior occurs, 
    the answer lies in the procedure followed
    to accomplish handing over the channel
    to the parking manager thread. This procedure
    is called masquerading. In the process,
    a duplicate copy of the channel is created,
    and most of the active data is given to the
    new copy. The original channel gets its name
    changed to XXX<ZOMBIE> and keeps the PBX
    information for the sake of the original
    thread (preserving its role as a call 
    originator, if it had this role to begin
    with), while the new channel is without
    this info and becomes a call target (a
    "peer").
    
    In this case, the parking lot manager
    thread is handed the new (masqueraded)
    channel. It will not run an h-exten
    on the channel if it hangs up while
    in the parking lot. The h exten will
    be run on the original channel instead,
    in the original thread, after the bridge
    completes.
    
    See bug 13820 for our intentions as
    to how to clean up the h exten behavior.
  
  Review: http://reviewboard.digium.com/r/29/
  
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-24 01:15:43 +00:00
Russell Bryant
52f5e8fcfb Merged revisions 165723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines

Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.

This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source.  While this usage was perfectly safe,
there are others that are problematic.  Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.

Further changes to get rid of KEEPALIVE and related code is being done by
murf.  There is a patch up for that on review 29.

Review: http://reviewboard.digium.com/r/98/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@165728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18 19:45:11 +00:00
Mark Michelson
ec71774eae Merged revisions 160626 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines

Add some safety measures when using gosub, especially when using the options
for app_dial and app_queue to run a gosub when the call is answered.

* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
  then this will cause errors when we attempt to actually run the gosub, including
  a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
  to actually run the gosub routine. If there was an error, we should not attempt
  to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.

(closes issue #13548)
Reported by: fiddur


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2008-12-03 18:42:01 +00:00
Mark Michelson
81826df05d Merged revisions 159554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines

Add some necessary hangup commands in the case that forwarding
a call fails

1) Hang up the original destination if the local channel cannot
   be requested.
2) Hang up the local channel (in addition to the original destination)
   if ast_call fails when calling the newly created local channel.

This prevents channels from sticking around forever in the
case of a botched call forward (e.g. to an extension which does not
exist).

(closes issue #13764)
Reported by: davidw
Patches:
      13764_v2.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, davidw


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@159561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-26 19:58:43 +00:00
Mark Michelson
7122ee6adb Merged revisions 158066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines

Merged revisions 158053 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines

Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.

(closes issue #13867)
Reported by: still_nsk
Patches:
      13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage


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2008-11-20 17:40:20 +00:00
Mark Michelson
892f98a1e6 Merged revisions 157306 via svnmerge from
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r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines

Merged revisions 157305 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines

Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.

This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.


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2008-11-18 18:32:54 +00:00
Tilghman Lesher
ed1113b08e Merged revisions 157253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r157253 | tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines
  
  Can't use items duplicated off the stack frame in an element returned from
  a function: in these cases, we have to use the heap, or garbage will result.
  (closes issue #13898)
   Reported by: alecdavis
   Patches: 
         20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14)
   Tested by: alecdavis
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2008-11-17 22:39:55 +00:00
Tilghman Lesher
1b8217193c Merged revisions 156388 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) | 12 lines
  
  Merged revisions 156386 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines
    
    When using call limits under 1 second, infinite call lengths are allowed,
    instead.
    (closes issue #13851)
     Reported by: ruddy
  ........
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2008-11-12 21:36:02 +00:00
Mark Michelson
0c0fdebd9f Merged revisions 156169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines

Merged revisions 156167 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines

When doing some tests, I was having a crash at the end of every call
if an attended transfer occurred during the call. I traced the cause to
the CDR on one of the channels being NULL. murf suggested a check in
the end bridge callback to be sure the CDR is non-NULL before proceeding,
so that's what I'm adding.


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2008-11-12 17:48:24 +00:00
Sean Bright
839cb83ea3 Merged revisions 155554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines

Merged revisions 155553 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines

Use static functions here instead of nested ones.  This requires a small
change to the ast_bridge_config struct as well.  To understand the reason
for this change, see the following post:

    http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-09 01:43:14 +00:00
Kevin P. Fleming
1036849a42 import gcc 4.3.2 warning fixes from trunk, with a few changes specific to this branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02 23:56:13 +00:00
Terry Wilson
239f1d53bc Merged revisions 153181 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines
  
  Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten.  Added a callback function to handle setting variables, etc. from w/in the bridging code.  Calls back into a nested function within the function calling ast_bridge_call
  
  (closes issue #13793)
  Reported by: greenfieldtech
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31 22:11:11 +00:00
Steve Murphy
039de710a7 Merged revisions 152605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r152605 | murf | 2008-10-28 23:47:13 -0600 (Tue, 28 Oct 2008) | 22 lines

Merged revisions 152538 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines

A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.

I hope this doesn't spoil some vast, eternal plan...


........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:52:41 +00:00
Steve Murphy
f1da1c3957 Merged revisions 152536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r152536 | murf | 2008-10-28 23:01:00 -0600 (Tue, 28 Oct 2008) | 57 lines

Merged revisions 152535 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines

The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the 
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.

If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.

If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.

Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden 
(in trunk).

All the places that previously tested for 
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.

I tested this against the 4 common parking
scenarios:


1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.

2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.

3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.

4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.


No crash.

I also ran the scenarios above against valgrind, and accesses looked good.



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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:09:20 +00:00
Tilghman Lesher
a7a57a4f61 Merged revisions 152369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines
  
  Merged revisions 152368 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines
    
    Reset all DIAL variables back to blank, in case Dial is called multiple times
    per call (which could otherwise lead to inconsistent status reports).
    (closes issue #13216)
     Reported by: ruddy
     Patches: 
           20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
     Tested by: ruddy
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-28 17:08:47 +00:00
Mark Michelson
28aafe781b Merged revisions 149279 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r149279 | mmichelson | 2008-10-14 18:57:46 -0500 (Tue, 14 Oct 2008) | 7 lines

When specifying an invalid timeout to Dial, take it
to mean that no timeout is desired.

(closes issue #13625)
Reported by: atis


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@149280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 23:58:10 +00:00
Sean Bright
b0d673495e Merged revisions 147050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r147050 | seanbright | 2008-10-07 08:01:36 -0400 (Tue, 07 Oct 2008) | 8 lines

Make sure to compare the correct number of characters when special-casing
our DAHDI operator mode stuff.  Technically, it would work fine, as 'DAH'
is currently unique amongst our channel technologies, but as Jared points
out:

  <@jsmith> Sure... as long as the technology starts whith DAH.... but
            it could be DAHDOO!

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@147052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-07 12:03:57 +00:00
Tilghman Lesher
25dd2fa2f3 Merged revisions 143031 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r143031 | tilghman | 2008-09-13 08:54:15 -0500 (Sat, 13 Sep 2008) | 8 lines
  
  Repair IAXVAR implementation so that it works again (regression?)
  (closes issue #13354)
   Reported by: adomjan
   Patches: 
         20080828__bug13354.diff.txt uploaded by Corydon76 (license 14)
         20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Corydon76, adomjan
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2008-09-13 13:58:15 +00:00
Steve Murphy
bc73329607 Merged revisions 142676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines

Merged revisions 142675 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines

Tested by: sergee, murf, chris-mac, andrew, KNK

This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from 
the ground up!

This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.

Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.

While I dearly hope that this patch overcomes all problems, and 
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.

** trunk note: some code to suppress the h exten being run 
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.


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2008-09-12 05:03:09 +00:00
Steve Murphy
41138ad333 Merged revisions 139627 via svnmerge from
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r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines

Merged revisions 139347 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines


(closes issue #13251)
Reported by: sergee
Tested by: murf



THis is a bold move for a static release fix, but I wouldn't have
made it if I didn't feel confident (at least a *bit* confident)
that it wouldn't mess everyone up.

The reasoning goes something like this:

1. We simply cannot do anything with CDR's at the current point
(in pbx.c, after the __ast_pbx_run loop). It's way too late to
have any affect on the CDRs. The CDR is already posted and gone,
and the remnants have been cleared.

2. I was very much afraid that moving the running of the 'h'
extension down into the bridge code (where it would be now
practical to do it), would result in a lot more calls to the
'h' exten, so I implemented it as another exten under another
name, but found, to my pleasant surprise, that there was a 
1:1 correspondence to the running of the 'h' exten in the
pbx_run loop, and the new spot at the end of the bridge.
So, I ifdef'd out the current 'h' loop, and moved it into
the bridge code. The only difference I can see is the stuff
about the AST_PBX_KEEPALIVE, and hopefully, if this 
is still an important decision point, I can replicate it
if there are complaints. To be perfectly honest,
the KEEPALIVE situation is not totally clear to me,
and how it relates to a post-bridge situation is less
clear. I suspect the users will point out everything
in total clarity if this steps on anyone's toes!

3. I temporarily swap the bridge_cdr into the channel
before running the 'h' exten, which makes it possible
for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set,
the users can also get at the billsec/duration vals.
After the h exten finishes, the cdr is swapped back
and processing continues as normal.

Please, all who deal with CDR's, please test this version
of Asterisk, and file bug reports as appropriate!


........

I also made a little fix to the app_dial's 'e' option,
that is related to my updates.


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2008-08-22 22:15:36 +00:00
Sean Bright
3ffb39833b More RSW merges. Everything from apps/ except for the big offenders
app_voicemail and app_queue.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-10 14:45:25 +00:00
Steve Murphy
5eaf8450d6 Merged revisions 135799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines

(closes issue #12982)
Reported by: bcnit
Tested by: murf

I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.

And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first 
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).

I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.

To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.

I also corrected one small mention of the Zap device
to equally consider the dahdi device.

I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.



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2008-08-05 23:45:32 +00:00
Kevin P. Fleming
7df8b8b848 make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05 16:56:11 +00:00
Mark Michelson
bd1bb0d0e2 Merged revisions 130792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines

Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-14 17:54:11 +00:00
Tilghman Lesher
da03cdd174 Merged revisions 129149 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines

Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
 Reported by: ibc
 Patches: 
       20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: ibc

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-08 20:30:29 +00:00
Kevin P. Fleming
da14954bdc another minor ast_channel memory size decrease... for nearly all channels, 'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-01 16:16:36 +00:00
Mark Michelson
0178d0ccd6 Improve consistency between app_dial and app_queue with regards
to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.

After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the 
previous behavior of app_dial if desired.

(closes issue #12489)
Reported by: bcnit



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-26 23:35:29 +00:00
Tilghman Lesher
90867b2b0c Channel lock janitor -- add locks around retrieval of channel variables
(closes issue #12840)
 Reported by: pputman
 Patches: 
       app_dial_threadsafe3.patch uploaded by pputman (license 81)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-18 13:09:02 +00:00
Steve Murphy
f4c85ebd22 (closes issue #12689)
Reported by: ys

Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.

I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c

I did a simple sanity test to make sure the code doesn't
mess things up in general.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-16 20:43:46 +00:00
Jeff Peeler
ef3b214728 Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12 17:27:55 +00:00