Commit Graph

16968 Commits

Author SHA1 Message Date
Jeff Peeler
ab660c7f77 Merged revisions 168517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168517 | jpeeler | 2009-01-12 15:51:46 -0600 (Mon, 12 Jan 2009) | 12 lines
  
  Merged revisions 168516 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines
    
    (closes issue #13881)
    Reported by: hoowa
    
    Update the app CDR field for AGI commands that are not executing an application via "exec".
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@168519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-12 22:00:32 +00:00
Jeff Peeler
76778300b1 Merged revisions 168508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168508 | jpeeler | 2009-01-12 14:53:04 -0600 (Mon, 12 Jan 2009) | 15 lines
  
  Merged revisions 168507 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines
    
    (closes issue #12269)
    Reported by: IgorG
    Tested by: denisgalvao
    
    This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock.
    
    Review: http://reviewboard.digium.com/r/35/
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2009-01-12 21:02:32 +00:00
Olle Johansson
b1a8c36bb0 Merged revisions 168497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168497 | oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines

Better to use the proper app name

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2009-01-12 17:26:38 +00:00
Mark Michelson
45340d2526 Merged revisions 168486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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2009-01-12 15:05:43 +00:00
Russell Bryant
2b5e595403 Merged revisions 168481 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168481 | russell | 2009-01-12 08:57:49 -0600 (Mon, 12 Jan 2009) | 10 lines

Merged revisions 168480 via svnmerge from 
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r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines

s/ringdance/ringcadence/ for Bulgaria

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2009-01-12 14:58:53 +00:00
Tilghman Lesher
02dd0d784f Merged revisions 168334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168334 | tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines
  
  sizeof for a stringfield is 4.  Kinda low for reconstructing a field value.
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2009-01-10 01:44:33 +00:00
Mark Michelson
6a018f161f Blocked revisions 168265 via svnmerge
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r168265 | mvanbaak | 2009-01-09 17:04:46 -0600 (Fri, 09 Jan 2009) | 9 lines

Add a script to find out the correct settings for Asterisk behind NAT

(closes issue #13065)
Reported by: tzafrir
Patches:
      sip_nat_settings uploaded by tzafrir (license 46)
      sip_nat_settings_6 uploaded by mvanbaak (license 7)
Tested by: tzafrir, pabelanger, Dovid and moi

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2009-01-09 23:23:59 +00:00
Kevin P. Fleming
337890ea22 Merged revisions 168270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan 2009) | 9 lines
  
  Merged revisions 168267 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line
    
    update to use new sound file packages that include license files
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2009-01-09 23:18:28 +00:00
Richard Mudgett
08c98706ee Merged revisions 168192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168192 | rmudgett | 2009-01-09 15:43:30 -0600 (Fri, 09 Jan 2009) | 10 lines
  
  Merged revisions 168191 via svnmerge from 
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    r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines
    
    *  Fix for JIRA AST-175/ABE-1757
    *  Miscellaneous doxygen comments added.
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2009-01-09 23:12:08 +00:00
Russell Bryant
ed7fbc7ea1 Merged revisions 168200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168200 | russell | 2009-01-09 16:21:05 -0600 (Fri, 09 Jan 2009) | 10 lines

Merged revisions 168198 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines

Make this compile for mvanbaak

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2009-01-09 22:23:57 +00:00
Mark Michelson
4736407e8d Merged revisions 168193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan 2009) | 21 lines

Merged revisions 168128 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines

Add check_via calls to more request handlers

INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.

(closes issue #13071)
Reported by: baron
Patches:
      check_via.patch uploaded by baron (license 531)
Tested by: baron

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2009-01-09 21:57:46 +00:00
Terry Wilson
0e782e9a28 Merged revisions 168142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168142 | twilson | 2009-01-09 14:25:25 -0600 (Fri, 09 Jan 2009) | 7 lines
  
  Don't leak memory if phoneprov.conf does not exist
  
  (closes issue #14203)
  Reported by: jamesgolovich
  Patches: 
        asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich (license 176)
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2009-01-09 20:30:21 +00:00
Tilghman Lesher
87d261b48e Merged revisions 168090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168090 | tilghman | 2009-01-09 12:30:55 -0600 (Fri, 09 Jan 2009) | 3 lines
  
  When using ast_str with a non-ast_str-enabled API, we need to update the buffer
  or otherwise, we cannot use ast_str_strlen().
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2009-01-09 18:42:16 +00:00
Matthew Nicholson
1f9def253b Merged revisions 168014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168014 | mnicholson | 2009-01-09 10:32:34 -0600 (Fri, 09 Jan 2009) | 5 lines
  
  Use ast_safe_system() in logger.c instead of system()
  
  (closes issue #14194)
  Reported by: pabelanger
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2009-01-09 16:41:28 +00:00
Terry Wilson
e33b84fca8 Blocked revisions 167973 via svnmerge
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  r167973 | twilson | 2009-01-08 19:15:43 -0600 (Thu, 08 Jan 2009) | 2 lines
  
  Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentation
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2009-01-09 01:17:00 +00:00
Terry Wilson
a671c82ec2 Merged revisions 167935 via svnmerge from
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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
Tilghman Lesher
f91ed59190 Merged revisions 167894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167894 | tilghman | 2009-01-08 16:37:20 -0600 (Thu, 08 Jan 2009) | 13 lines
  
  Merged revisions 167840 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines
    
    Don't truncate database results at 255 chars.
    (closes issue #14069)
     Reported by: evandro
     Patches: 
           20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
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2009-01-08 22:45:36 +00:00
Mark Michelson
dae0c0060d Blocked revisions 167792,167888 via svnmerge
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r167792 | mmichelson | 2009-01-08 13:48:42 -0600 (Thu, 08 Jan 2009) | 15 lines

Add the average talk time for a queue

This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.

(closes issue #13960)
Reported by: coolmig
Patches:
      app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)


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r167888 | mmichelson | 2009-01-08 16:34:52 -0600 (Thu, 08 Jan 2009) | 4 lines

Revert chan_sip changes which were accidentally committed
in revision 167792


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2009-01-08 22:37:02 +00:00
Tilghman Lesher
77bf104602 Blocked revisions 167837 via svnmerge
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  r167837 | tilghman | 2009-01-08 15:40:28 -0600 (Thu, 08 Jan 2009) | 2 lines
  
  Fix variables to comply with documentation changes
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Tilghman Lesher
361d76c1eb Merged revisions 167835 via svnmerge from
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  r167835 | tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines
  
  Textual changes, consistency in status variable naming, and other minor bugs.
  (closes issue #13943)
   Reported by: Marquis
   Patches: 
         minivm_trunk_fixes3.patch uploaded by Marquis (license 32)
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2009-01-08 21:39:31 +00:00
Kevin P. Fleming
b205a18a8d Merged revisions 167720 via svnmerge from
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  r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan 2009) | 9 lines
  
  Merged revisions 167714 via svnmerge from 
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    r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line
    
    remove an unnecessary argument to queue_request()
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2009-01-08 17:28:42 +00:00
Kevin P. Fleming
4f2b0aa24c Merged revisions 167700 via svnmerge from
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  r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan 2009) | 12 lines
  
  Merged revisions 167620 via svnmerge from 
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    r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines
    
    When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available.
    
    http://reviewboard.digium.com/r/123/
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2009-01-08 16:59:07 +00:00
Leif Madsen
39f02645de Merged revisions 167662 via svnmerge from
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  r167662 | lmadsen | 2009-01-08 09:27:53 -0500 (Thu, 08 Jan 2009) | 1 line
  
  Oops... fix the fieldname I changed yesterday to be right.
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2009-01-08 14:30:54 +00:00
Russell Bryant
424dc6af48 Merged revisions 167569 via svnmerge from
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r167569 | russell | 2009-01-07 16:36:34 -0600 (Wed, 07 Jan 2009) | 10 lines

Merged revisions 167566 via svnmerge from 
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r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines

Fix the last couple of places where free() was improperly used directly.

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Russell Bryant
e0d2263f33 Merged revisions 167555 via svnmerge from
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r167555 | russell | 2009-01-07 16:27:23 -0600 (Wed, 07 Jan 2009) | 10 lines

Merged revisions 167554 via svnmerge from 
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r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines

Don't fclose() the file early, the filestream destructor will handle it.

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2009-01-07 22:28:09 +00:00
Russell Bryant
fcca2ae242 Merged revisions 167546 via svnmerge from
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r167546 | russell | 2009-01-07 16:20:31 -0600 (Wed, 07 Jan 2009) | 10 lines

Merged revisions 167545 via svnmerge from 
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r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines

Only try to close the file if one was actually opened

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2009-01-07 22:21:40 +00:00
Russell Bryant
e09314f28c Merged revisions 167542 via svnmerge from
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r167542 | russell | 2009-01-07 16:05:29 -0600 (Wed, 07 Jan 2009) | 12 lines

Merged revisions 167541 via svnmerge from 
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r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines

Don't use free() directly.  This caused a crash since ast_filestream is now an ao2 object.

Reported by JunK-Y on IRC, #asterisk-dev

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BJ Weschke
32be29070a Merged revisions 167478 via svnmerge from
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  r167478 | bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines
  
   Answer the channel if it has not already been answered and we've already found a valid profile for followme. 
   (closes issue #14140)
   Reported by: dimas
   Patches:
         14140.patch uploaded by dimas
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2009-01-07 18:32:52 +00:00
Leif Madsen
12d06b75b4 Merged revisions 167477 via svnmerge from
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  r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 Jan 2009) | 8 lines
  
  Update queues.conf.sample documentation.
  
  Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
  
  
  (closes issue #14179)
  Reported by: CrashHD
  Tested by: CrashHD
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2009-01-07 18:27:35 +00:00
Russell Bryant
b7d3a078c3 Merged revisions 167442 via svnmerge from
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r167442 | russell | 2009-01-07 11:35:39 -0600 (Wed, 07 Jan 2009) | 12 lines

Merged revisions 167432 via svnmerge from 
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r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) | 4 lines

Treat an empty string the same way as a NULL country argument.

In passing, simplify the handling of returning a default tone zone.

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2009-01-07 17:46:02 +00:00
Leif Madsen
091e03e136 Merged revisions 167373 via svnmerge from
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  r167373 | lmadsen | 2009-01-07 09:26:19 -0500 (Wed, 07 Jan 2009) | 1 line
  
  Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is
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2009-01-07 14:41:14 +00:00
Mark Michelson
ee8553caa6 Merged revisions 167301 via svnmerge from
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r167301 | mmichelson | 2009-01-06 15:36:44 -0600 (Tue, 06 Jan 2009) | 16 lines

Merged revisions 167299 via svnmerge from 
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r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) | 8 lines

Use the correct variable when creating the format string

(closes issue #14177)
Reported by: nic_bellamy
Patches:
      asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299)


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Tilghman Lesher
c7add836c6 Merged revisions 167265 via svnmerge from
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  r167265 | tilghman | 2009-01-06 15:02:33 -0600 (Tue, 06 Jan 2009) | 16 lines
  
  Merged revisions 167260 via svnmerge from 
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    r167260 | tilghman | 2009-01-06 14:48:05 -0600 (Tue, 06 Jan 2009) | 9 lines
    
    Merged revisions 167259 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.2
    
    ........
      r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines
      
      Security fix AST-2009-001.
    ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@167268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-06 21:10:53 +00:00
Mark Michelson
0e408f80fe Merged revisions 167180 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan 2009) | 49 lines

Merged revisions 167179 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines

A couple of changes to T.38 SDP attribute handling

There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and 
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.

It was also discovered that a particular type of 
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.

Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.

(closes issue #13976)
Reported by: linulin
Patches:
     chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
	 chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@167182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-05 17:10:08 +00:00
Tilghman Lesher
cfd27e5c80 Merged revisions 167176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r167176 | tilghman | 2009-01-05 10:44:47 -0600 (Mon, 05 Jan 2009) | 7 lines
  
  More clearly explain that quote marks are no longer necessary.
  (closes issue #13718)
   Reported by: davidw
   Patches: 
         20081020__bug13718.diff.txt uploaded by Corydon76 (license 14)
   Tested by: blitzrage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@167178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-05 16:46:28 +00:00
Jeff Peeler
3190dfc8c4 Blocked revisions 167125 via svnmerge
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r167125 | jpeeler | 2009-01-03 14:29:54 -0600 (Sat, 03 Jan 2009) | 3 lines

When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.


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2009-01-03 20:32:35 +00:00
Terry Wilson
978a8ed0ac Blocked revisions 167057 via svnmerge
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  r167057 | twilson | 2008-12-31 16:53:55 -0600 (Wed, 31 Dec 2008) | 2 lines
  
  Don't forget to free typename
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@167058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31 22:55:11 +00:00
Mark Michelson
08d772b888 Blocked revisions 167021 via svnmerge
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r167021 | mmichelson | 2008-12-31 15:52:02 -0600 (Wed, 31 Dec 2008) | 4 lines

Change some incorrect syntax for pri set debug and correct
an off-by-one error in ss7 set debug command


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2008-12-31 21:54:33 +00:00
Tilghman Lesher
27c0f99cb4 Merged revisions 166954 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166954 | tilghman | 2008-12-31 13:34:28 -0600 (Wed, 31 Dec 2008) | 12 lines
  
  Merged revisions 166953 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines
    
    Also inherit the musiconhold class.
    (Closes #14153)
    Reported by: Jerry Geis, via the users list.
    Patch by: me (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31 19:38:10 +00:00
Terry Wilson
1c84ecad28 Merged revisions 166908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166908 | twilson | 2008-12-30 14:50:05 -0600 (Tue, 30 Dec 2008) | 2 lines
  
  Fix some svn:keywords
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-30 20:57:15 +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
Joshua Colp
75d4f7e1c2 Per kpfleming add a note describing why you must never change the first element of peer_finding_info.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29 14:52:22 +00:00
Eliel C. Sardanons
f2a3836106 Blocked revisions 166823 via svnmerge
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  r166823 | eliel | 2008-12-28 10:36:25 -0500 (Sun, 28 Dec 2008) | 3 lines
  
  Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-28 15:41:44 +00:00
Russell Bryant
941a82916a Merged revisions 166773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166773 | russell | 2008-12-28 09:15:14 -0600 (Sun, 28 Dec 2008) | 12 lines

Merged revisions 166772 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines

Use strncat() instead of an sprintf() in which source and target buffers overlap

http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html

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2008-12-28 15:16:11 +00:00
Terry Wilson
869388718f Blocked revisions 166731 via svnmerge
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  r166731 | twilson | 2008-12-24 09:10:42 -0600 (Wed, 24 Dec 2008) | 10 lines
  
  There is no section 22.2.2 in rfc 3261.  I believe 26.2.2 is what was meant:
  
        Note that in the SIPS URI scheme, transport is independent of TLS,
        and thus "sips:alice@atlanta.com;transport=tcp" and
        "sips:alice@atlanta.com;transport=sctp" are both valid (although
        note that UDP is not a valid transport for SIPS).  The use of
        "transport=tls" has consequently been deprecated, partly because
        it was specific to a single hop of the request.  This is a change
        since RFC 2543.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-24 15:19:33 +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
Tilghman Lesher
827f655dac Merged revisions 166696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines
  
  Allow semicolons and extended characters in user-specified SIP headers.
  (closes issue #14110)
   Reported by: gork
   Patches: 
         20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14)
   Tested by: gork, putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 20:56:10 +00:00
Mark Michelson
63f2d52453 Merged revisions 166569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec 2008) | 20 lines

Merged revisions 166568 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines

Fix a crash resulting from a datastore with inheritance but no duplicate callback

The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.

(closes issue #14113)
Reported by: francesco_r
Patches:
      14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r


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2008-12-23 15:20:15 +00:00
Tilghman Lesher
9981739dd1 Merged revisions 166533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) | 11 lines
  
  Merged revisions 166509 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines
    
    Use the integer form of condition for integer comparisons.
    (closes issue #14127)
     Reported by: andrew
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 04:34:13 +00:00
Mark Michelson
2ff6f18737 Merged revisions 166470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r166470 | mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11 lines

Always use the value of the AGISIGHUP when running an AGI.

Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel.

(closes issue #13711)
Reported by: fmueller
Patches:
      13711.patch uploaded by putnopvut (license 60)


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2008-12-22 23:27:37 +00:00