malloc() with an ast_free (which, of course, doesn't match up with known
allocated memory, so the free fails).
(closes issue #13702)
 Reported by: eliel
 Patches: 
       codec_lpc10_lpcini.c uploaded by eliel (license 64)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13697)
 Reported by: alecdavis
 Patches: 
       app_echo.bug13697.103249.diff.txt uploaded by alecdavis (license 585)


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We're going to try to get time to fix this and kpfleming believes that there's code in ao2 
so that we can solve it...


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct 2008) | 3 lines
  
  fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support
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reverting this change... had not read the commit list yet, didn't realize the code had been upgraded

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to mean that no timeout is desired.

(closes issue #13625)
Reported by: atis



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct 2008) | 4 lines

Change this warning to an error message. Suggestion
comes from Sean Bright. Thanks Sean!


........


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r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines

Call register_peer_exten even in the case that the peer's
IP/port does not change.

(closes issue #13309)
Reported by: dimas
Patches:
      v2-13309.patch uploaded by dimas (license 88)


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r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines

Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.

Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet


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r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines

Update the queue with the correct number of calls and
whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.

(closes issue #13395)
Reported by: Marquis
Patches:
      app_queue.c.transfer.patch uploaded by Marquis (license 32)


........


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leaks found with these changes.
(Closes issue #13505, closes issue #13543)
Reported by: mav3rick, triccyx
 Patches: 
       20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
 Tested by: mav3rick, triccyx


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r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines

Don't allow reserved characters to be used in register
lines in sip.conf.

(closes issue #13570)
Reported by: putnopvut


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  r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines
  
  Check correct values in the return of ast_waitfor(); also, get rid of a
  possible memory leak.
  (closes issue #13658)
   Reported by: explidous
   Patch by: me
........


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(closes issue #13342)
Reported and patch by: Laureano



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  r148990 | lmadsen | 2008-10-14 15:05:12 -0400 (Tue, 14 Oct 2008) | 4 lines
  
  Add in some missing updates to the CHANGES file for sip.conf
  (closes issue #13100)
  Reported and patch by: gknispel_proformatique
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  r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines
  
  Some compilers warn, some don't.  Fixing.
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(closes issue #13617)
 Reported by: alecdavis
 Patches: 
       app_sms.13oct.diff.txt uploaded by alecdavis (license 585)


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  r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines
  
  Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used
  in headers like 'Subject' and 'To'.
  Closes AST-107.
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r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct 2008) | 9 lines

Deadlock prevention in chan_local.

(closes issue #13676)
Reported by: tacvbo
Patches:
      13676.patch uploaded by putnopvut (license 60)
Tested by: tacvbo


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(closes issue #13688)
 Reported by: irroot
 Patches: 
       app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix reference count issue that Russell brought up in SIP MWI NOTIFY support. Bump the reference count up before we add it to the scheduler, duh.


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  r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct 2008) | 3 lines
  
  on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly
........


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ensure that *all* fields in the req structure are cleared out before reusing it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled


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  r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines
  
  it would be nice if this message printing code had actually been tested before it was committed...
........


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'update2', which permits updates which match across multiple columns, instead
of requiring all tables to have a single unique identifier.  All of the other
API calls with the exception of 'update' already had the ability to match on
multiple fields, so it was a missing and very desireable feature that an API
call implementing an update should have this, too.

This does not change any outward performance of Asterisk, but it should make
life easier for application developers who use the RealTime framework.


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the trie info when they do 'dialplan show ...'
(even with debug set to non-zero); so I set up a 
   'dialplan debug [context]' cli command instead, 
to explicitly show just the trie info.  I even
added an extension_exists() call to make sure the
trie info is built. I moved the explanatory header
to above the extension loop to ensure it only prints
once. And it will do this now, whether debug is set
or not.

I removed the trie printing from the 'dialplan show' 
command entirely. 



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- Fixing language


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(found at SIPit)


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(closes issue #13666)
Reported by: eliel
Patches: 
      res_agi.c.patch uploaded by eliel (license 64)


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in the "kill the user" commit and caused calls relying
on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that 
commit.

(closes issue #13644)
Reported by: pj
Patches:
      13644_trunkv2.patch uploaded by putnopvut (license 60)
Tested by: pj



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a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well.

(closes issue #13668)
Reported by: mjc



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kpfleming).


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