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

........
  r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
  
  Resolve a crash related to a T.38 reinvite race condition.
  
  This change resolves a crash observed locally during some T.38 testing.
  A call was set up using a call file, and when the T.38 reinvite came in,
  the channel state was still AST_STATE_DOWN.  The reason is explained by
  a comment in the code that previously lived in the handling of
  AST_STATE_RINGING.  This change modifies the logic to handle the same
  race condition for any channel state that is not UP.
  
  (closes ABE-1895)
........


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

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  r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
  
  Improved chan_dahdi.conf pritimer error checking.
  
  Valid format is: pritimer=timer_name,timer_value
  
  *  Fixed segfault if the ',' is missing.
  *  Completely check the range returned by pri_timer2idx() to prevent
  possible access outside array bounds.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines
  
  Use the handy UNLINK macro instead of hand-coding the same thing in-line.
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also change the preferred configuration option from 'hostname' (which was
misleading because it didn't actually treat the value as a hostname) to
'connection' and added some verbage explaining that the user would need to
refer to their freetds.conf file for those settings.  'hostname' was kept
as a backwards compatible configuration parameter.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202840 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202761 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the outgoing_colp parameter is set to not send COLP information, then
it does not make sense to send redirecting or transfer messages announcing
new COLP information that is blocked.  The service provider may supply the
listed number for that line when it passes the messages to the next hop.
Why tell the switch that these events happened when the information is
otherwise suppressed?

Also blocked the number of previous redirects that may have occurred to
calls going out the port when outgoing_colp is 2.

Follow on to JIRA ABE-1853.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14509)
Reported by: timeshell
Patches:
      20090504__bug14509.diff.txt uploaded by tilghman (license 14)
Tested by: awk, timeshell



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202753 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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  r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines
  
  MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport
  
  (closes issue #14659)
  Reported by: klaus3000
  Patches:
        patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65)
        mwi_port-transport_trunk.diff uploaded by dvossel (license 671)
  Tested by: dvossel, klaus3000
  
  Review: https://reviewboard.asterisk.org/r/288/
........


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........
  r202601 | mmichelson | 2009-06-23 10:22:35 -0500 (Tue, 23 Jun 2009) | 3 lines
  
  Fix more memory leaks that may result if rtp is not successfully allocated.
........


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........
  r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines
  
  Fix potential memory leak in chan_sip when video rtp is not allocated properly.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #2264)
Reported by: pfn
Patches:
      silent-vm-1.6.2.txt uploaded by pfn (license 810)


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

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  r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines
  
  Report CallerID change during a masquerade.
  
  Reported by: markster
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Pointed out by Russell while working on the CEL branch.


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

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  r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines
  
  Make Polycom subscription type override check more explicit.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202415 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Modules placed in the priority heap for loading were not properly removed from the linked list.  This resulted in some modules attempting to load twice.



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

........
  r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines
  
  Fix a situation in which Asterisk would not stop retransmitting 487s.
  
  If a CANCEL were received by Asterisk, we would send a 487 in response
  to the original INVITE and a 200 OK for the CANCEL. If there were a network
  hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating
  with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used
  to be to try sending another 487 to the canceled INVITE and another 200 OK to the
  CANCEL.
  
  The problem here is that the originally-sent 487 was sent "reliably" meaning that
  it will be retransmitted until it is received properly. So when we receive the second
  CANCEL it is likely that the first batch of 487s we sent is still going strong and
  reaches the UA. The result was that the second set of 487s would be retransmitted
  constantly until the maximum number of retries had been reached.
  
  The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel
  the retransmission of the first set of 487s and start a second set. This causes the
  dialog to be terminated reasonably.
  
  (closes issue #14584)
  Reported by: klaus3000
  Patches:
        14584_v2.patch uploaded by mmichelson (license 60)
  Tested by: klaus3000
........
  r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines
  
  Remove an extra debug line left from previous commit.
........


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

........
  r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines
  
  Fix a possible infinite loop in SDP parsing during glare situation.
  
  There was a while loop in get_ip_and_port_from_sdp which was controlled
  by a call to get_sdp_iterate. The loop would exit either if what we were
  searching for was found or if the return was NULL. The problem is that
  get_sdp_iterate never returns NULL. This means that if what we were searching
  for was not present, the loop would run infinitely. This modification of the
  loop fixes the problem.
  
  (closes issue #15213)
  Reported by: schmidts
  
  (closes issue #15349)
  Reported by: samy
  
  (closes issue #14464)
  Reported by: pj
  
  (closes issue #15345)
  Reported by: aragon
  Patches:
        sip_inf_loop.patch uploaded by mmichelson (license 60)
  Tested by: aragon
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202262 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15355)
Reported by: deuffy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In passing, also tweak cdr_unregister() to hold the list lock a bit less time.


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........
  r202022 | mnicholson | 2009-06-19 16:21:15 -0500 (Fri, 19 Jun 2009) | 4 lines
  
  Added deadlock protection to try_suggested_sip_codec in chan_sip.c.
  
  Review: https://reviewboard.asterisk.org/r/287/
........


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

........
  r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines
  
  timestamp was being converted to host order as a short rather than a long
  
  (closes issue #15361)
  Reported by: ffloimair
  Patches:
        ts_issue.diff uploaded by dvossel (license 671)
........


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201944 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reported by Benny Amorsen via the asterisk-users mailing list.
http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance.


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

........
  r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines
  
  If the "h" extension fails, give it another chance in main/pbx.c.
  If the "h" extension fails, give it another chance in main/pbx.c, when it
  returns from the bridge code.  Fixes an issue where the "h" extension may
  occasionally not fire, when a Dial is executed from a Macro.
  Debugged in #asterisk with user tompaw.
........


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within app_voicemail for directory functions.  It is therefore no longer
necessary for app_directory to be linked against the ODBC libraries (and it
never was necessary for app_directory to be linked against IMAP, though it
was).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15320)
 Reported by: chappell
 Patches: 
       cut_fix.patch uploaded by chappell (license 8)
       cut_clarify.patch uploaded by chappell (license 8)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201745 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/285/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #15269)
Reported by: contactmayankjain
Patches:
      patch.txt uploaded by contactmayankjain (license 740)
      memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel




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

........
  r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines
  
  Fix memory corruption and leakage related reloads of non files mode MoH classes.
  
  For Music on Hold classes that are not files mode, meaning that we are executing
  an application that will feed us audio data, we use a thread to monitor the
  external application and read audio from it.  This thread also makes use of the
  MoH class object.  In the MoH class destructor, we used pthread_cancel() to ask
  the thread to exit.  Unfortunately, the code did not wait to ensure that the
  thread actually went away.  What needed to be done is a pthread_join() to ensure
  that the thread fully cleans up before we proceed.  By adding this one line, we
  resolve two significant problems:
  
    1) Since the thread was never joined, it never fully goes away.  So, on every
       reload of non-files mode MoH, an unused thread was sticking around.
  
    2) There was a race condition here where the application monitoring thread
       could still try to access the MoH class, even though the thread executing
       the MoH reload has already destroyed it.
  
  (issue #15109)
  Reported by: jvandal
  
  (issue #15123)
  Reported by: axisinternet
  
  (issue #15195)
  Reported by: amorsen
  
  (issue AST-208)
........


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