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

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  r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
  
  Randomize IAX2 encryption padding
  
  The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all.  This patch calls ast_random to fill the padding buffer with random data.  The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
  
  Review: http://reviewboard.digium.com/r/193/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution.
Previously, FILE() returned one less character than specified, due to the
terminating NULL.  Both the offset and length parameters now behave
identically to the way variable substitution offsets and lengths also work.
(closes issue #14670)
 Reported by: BMC


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

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  r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines
  
  Fixup glare detection, to fix a memory leak of a local pvt structure.
  (closes issue #14656)
   Reported by: caspy
   Patches: 
         20090313__bug14656__2.diff.txt uploaded by tilghman (license 14)
   Tested by: caspy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For a channel that is not yet answered this API call will wait
until a voice frame is received on the channel before returning.
It does this by waiting for frames on the channel and reading them
in. The frames read in were not freed when they should have been.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182171 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14227)
Reported by: caspy



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

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  r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines
  
  Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF.
  
  Dynamic features defined in the applicationmap section of features.conf allow
  one to specify whether the caller, callee, or both have the ability to use the
  feature. The documentation in the features.conf.sample file could be interpreted
  to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the
  calling channel in order to allow for the callee to be able to use the features
  which he should have permission to use. However, the DYNAMIC_FEATURES variable
  would only be read from the channel of the participant that pressed the DTMF
  sequence to activate the feature. The result of this was that the callee was
  unable to use dynamic features unless the dialplan writer had taken measures
  to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel.
  
  This commit changes the behavior of ast_feature_interpret to concatenate the
  values of DYNAMIC_FEATURES from both parties involved in the bridge. The features
  themselves determine who has permission to use them, so there is no reason to believe
  that one side of the bridge could gain the ability to perform an action that they
  should not have the ability to perform.
  
  Kevin Fleming pointed out on the asterisk-users list that the typical way that this
  was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel
  so that the value would be inherited by the called channel. While this works, the
  documentation alone is not enough to figure out why this is necessary for the callee
  to be able to use dynamic features. In this particular case, changing the code to match
  the documentation is safe, easy, and will generally make things easier for people for
  future installations.
  
  This bug was originally reported on the asterisk-users list by David Ruggles.
  
  (closes issue #14657)
  Reported by: mmichelson
  Patches:
        14657.patch uploaded by mmichelson (license 60)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The code responsible for sending the T38 reinvite did not check if an INVITE was
already being handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current INVITE is done being
handled.

(issue AST-191)


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

Just recording the v1.4 change in trunk since it originally came from here.

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  r181898 | rmudgett | 2009-03-12 20:19:29 -0500 (Thu, 12 Mar 2009) | 4 lines
  
  Use the correct branch integrated property when generating the version string.
  
  Copied the make_version file from Asterisk trunk.
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  r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
  
  Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
  
  If we receive an INVITE from an endpoint and then later receive a BYE from that
  same endpoint before we have sent a final response for the INVITE, then we need
  to respond to the INVITE with a 487. 
  
  There was logic in the code prior to this commit which seemed to exist solely to 
  handle this situation, but there was one condition in an if statement which 
  was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
  channel. This made no sense since we created the owner channel when we received
  the INVITE, meaning that the majority of the time we would never send the 487.
  The 487 being sent should not rely on whether we have created a channel. Its
  delivery should be dependent on the current state of the initial INVITE transaction.
  With this commit, that logic is now correctly in place.
  
  (closes issue #14149)
  Reported by: legranjl
  Patches:
        14149.patch uploaded by mmichelson (license 60)
  Tested by: legranjl
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Previously, only 5 columns were displayed, and if a translation time exceeded
99,999 useconds, it would be displayed as 0, instead of its actual time.
(closes issue #14532)
 Reported by: pj
 Patches: 
       20090311__bug14532.diff.txt uploaded by tilghman (license 14)
 Tested by: pj


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  r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines
  
  Fix incorrect usage of strncasecmp... I really meant to use strcasecmp.
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  r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines
  
  Fix another scenario where depending on configuration the stream would not get read.
  
  For custom commands we don't know whether the audio is coming from a stream or not
  so we are going to have to read the data despite no channels.
  
  (closes issue #14416)
  Reported by: caspy
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  r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines
  
  Fix logic flaw in previous commit.
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  r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines
  
  Fix issue with streaming MOH failing if nobody is listening.
  
  When a music class is setup to actually provide music on hold
  from a stream we need to constantly read audio from it since it
  will constantly be providing audio. This is now done despite there
  being no channels listening to it.
  
  (closes issue #14416)
  Reported by: caspy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14647)
Reported by: sherpya


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(closes issue #14646)
Reported by: strk


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  r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines
  
  Allow prefix to set localstatedir (when used and different from the default).
  
  This is similar to the /etc change that was made for the non-FreeBSD case.
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines

Make code that updates BRIDGEPEER variable thread-safe.

It is not safe to read the name field of an ast_channel without the channel
locked.  This patch fixes some places in channel.c where this was being done,
and lead to crashes related to masquerades.

(closes issue #14623)
Reported by: guillecabeza

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  r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
  
  encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
  
  If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted.  This causes the entire frame to be corrupted.  When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense.  When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop.  To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted.  Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
  
  (closes issue #14607)
  Reported by: stevenla
  Tested by: dvossel
  
  Review: http://reviewboard.digium.com/r/192/
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  r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
  
  Fix issue where an attended transfer could not be completed under a rare scenario.
  
  When completing an attended transfer chan_sip does a check to make sure the extension
  in the URI portion of the Refer-To header is a local valid extension. We don't actually
  need to check this since we know for sure the other channel is already up and talking to
  the extension. Some devices do not put the extension in the Refer-To header either, which
  can cause the extension check to fail. We now no longer do this check if it is an attended
  transfer.
  
  (closes issue #14628)
  Reported by: sverre
  Patches:
        14628.diff uploaded by file (license 11)
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
  
  Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
  
  When dtmfmode was set to auto the inband DTMF detector was not setup
  on outgoing SIP calls. This caused inband DTMF detection to fail.
  The inband DTMF detector is now setup for both dtmfmode inband and auto.
  
  (closes issue #13713)
  Reported by: makoto
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If trying to dial a non-existent queue, there would
be a segfault when attempting to access q->weight, even
though q was NULL. This problem was introduced during
the queue-reset merge and thus only affects trunk.

(closes issue #14643)
Reported by: alecdavis



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181244 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet
option is enabled. (reported by Vadim Lebedev on the asterisk-dev list)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181210 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue. 

A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.

(closes issue #13593)
Reported by: pj



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  r181133 | jpeeler | 2009-03-10 22:25:04 -0500 (Tue, 10 Mar 2009) | 13 lines
  
  Fix malloc debug macros to work properly with h323.
  
  The main problem here was that cstdlib was undefining free thereby causing the
  proper debug macros to not be used. ast_h323.cxx has been changed to call
  ast_free instead to avoid the issue. Because using the ast prefix calls are
  a better choice, ast_free_ptr is the new wrapper for free to pass to functions.
  Also, a little bit of clean up was done to avoid the debug macros intentionally
  being redefined.
  
  (closes issue #13593)
  Reported by: pj
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines
  
  Fix incorrect tag checking on transfers when pedantic=yes is enabled.
  
  (closes issue #14611)
  Reported by: klaus3000
  Patches:
        patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65)
  Tested by: klaus3000
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  r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines
  
  Remove unused variables.
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(related to issue #14636)


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  r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line
  
  Make things happier when using autoconf 2.62+
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