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  r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines
  
  Merged revisions 283558 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines
    
    Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
    
    Asterisk now dynamically builds the "Supported" header depending
    on what is enabled/disabled in sip.conf.  Session timers used
    to always be advertised as being supported even when they were disabled
    in the configuration.  This caused problems with some end points.
    
    (issue #17005)
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  r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines
  
  Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
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(closes issue #17887)
Reported by: salecha
Patches:
      skinny.redial.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha


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  r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines
  
  Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
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  r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines
  
  Fix issue where TOS is no longer set on RTP packets.
  Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk.
  
  (closes issue #17890)
  Reported by: elguero
  Patches:
        qos_18.diff uploaded by elguero (license 37)
  
  Review: https://reviewboard.asterisk.org/r/868
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  r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines
  
  Merged revisions 283381 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines
    
    Merged revisions 283380 via svnmerge from 
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    ........
      r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines
      
      This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
      
      When the pending bye flag is used, it is possible that the dialog will terminate
      and leave the sip_pvt->owner channel up.  This is because we never hangup the
      ast_channel after sending the SIP_BYE request.  When we receive the response for
      the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the
      next do_monitor loop, but this is not the case.  The dialog will only be destroyed
      once the owner is hungup even with the need_destroy flag set.  This patch sets the
      softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the
      pending bye flag.
    ........
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  r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) | 2 lines
  
  Don't attempt to release a NULL ODBC handle.
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  r283319 | tilghman | 2010-08-23 16:33:47 -0500 (Mon, 23 Aug 2010) | 9 lines
  
  Merged revisions 283318 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010) | 2 lines
    
    CDR drivers depend upon res_odbc, not directly on the ODBC libraries
  ........
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Also correct the callinfo provided in skinny_answer.

(closes issue #17876)
Reported by: salecha
Patches:
      skinny_cnd3.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn

Review: NA 


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  r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23 Aug 2010) | 2 lines
  
  Add sample configuration for cel_radius.
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  r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines
  
  Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
  
  Really, having 2 enums for this is silly and error prone, demonstrated by
  the crash that I hit because there was an assumption in the code that the
  values in each matched up.  However, this is a quick fix to get them to
  match up so it will work.
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  r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) | 2 lines
  
  Don't blow up on an invalid AMA flag.
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  r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010) | 2 lines
  
  Tack on ${eventextra} to the sample cel_custom.conf.
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  r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010) | 2 lines
  
  Cut down on excessive quotation.
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  r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 Aug 2010) | 2 lines
  
  Don't fail to start if the config file is missing.
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  r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) | 5 lines
  
  Expand cel_custom.conf.sample.
  
  Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting.  Also list
  the special CEL variables that are available for use in the mapping.
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  r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines
  
  Merged revisions 283049 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines
    
    Merged revisions 283048 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines
      
      Q931 - Sending PROGRESS after sending ALERTING is a protocol error
      
      The PRI layer in chan_dadhi will check if a PROGRESS message has already
      been sent, and not allow sending another (although that is technically
      allowed by the Q931 spec), however it does not protect against sending an
      ALERTING and then sending a PROGRESS message, which is a violation of the
      specification.
      
      Most switches don't seem to care too deeply about this, but some do, and
      will disconnect the call when receiving this invalid sequence.
      
      Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure
      A.5/Q.931 -- Overview protocol control (network side) point-point
      (sheet 3 of 8)"
      
      (closes issue #17874)
      Reported by: nic_bellamy
      Patches:
            asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
            asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
            asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
    ........
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  r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010) | 2 lines
  
  Fix a typo in a column name.
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  r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010) | 2 lines
  
  Add an argument missing from the CELGenUserEvent documentation.
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  r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) | 4 lines
  
  Split _all_ arguments before parsing them.
  
  This fixes multicast RTP paging using linksys mode.
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  r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines
  
  Merged revisions 282894 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines
    
    Merged revisions 282893 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines
      
      tos_sip option was not being set correctly
      
      When tos_sip is used, the tos of the sip socket is only set
      correctly if the socket binding changes on a reload.  If the binding
      stays the same but the TOS changes, the new tos value would not take
      into effect.  This patch fixes that.
      
      
      (closes issue #17712)
      Reported by: nickb
    ........
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  r282891 | dvossel | 2010-08-19 15:34:41 -0500 (Thu, 19 Aug 2010) | 11 lines
  
  Merged revisions 282890 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines
    
    fixes sip peer memory leaks in the peer_by_ip table
    
    (issue #17798)
  ........
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  r282860 | mnicholson | 2010-08-19 15:01:11 -0500 (Thu, 19 Aug 2010) | 30 lines
  
  Merged revisions 282859 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines
    
    Merged revisions 277944 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines
      
      Regression with T.38 negotiation
      
      Prior to 1.4.26.3 T.38 negotiation worked properly, in the case
      of the reporter.  
      
      (issue #16852)
      Reported by: cfc
      
      (closes issue #16705)
      Reported by: mpiazzatnetbug
      Patches:
            issue16705_2.diff uploaded by ebroad (license 878)
      Tested by: vrban, ebroad, c0rnoTa, samdell3
      
      Review: https://reviewboard.asterisk.org/r/754/
    ........
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  r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) | 2 lines
  
  Only output debugging if the debug level is on.
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  r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines
  
  Merged revisions 282730 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines
    
    Merged revisions 282729 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
      
      Add some documentation about codec negotiation to sip.conf
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Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one.

(closes issue #17812)
Reported by: wedhorn
Patches:
      cleanup.stateoffhook.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn

Review: NA 


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  r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) | 1 line
  
  Use the correct operator when calculating the PRI span devstate.
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  r282672 | rmudgett | 2010-08-18 10:28:27 -0500 (Wed, 18 Aug 2010) | 1 line
  
  Use the correct type for aoce_delayhangup bit field.
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  r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines
  
  Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests.
  
  This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests.  These changes to NOTIFY handler were first introduced in r217482.  This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received.
  
  (issue #17486)
  Reported by: davidw
  Tested by: mnicholson
  
  (issue #12713)
  Reported by: davidw
  
  Review: https://reviewboard.asterisk.org/r/860/
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  r282608 | tilghman | 2010-08-18 02:49:04 -0500 (Wed, 18 Aug 2010) | 16 lines
  
  Merged revisions 282607 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines
    
    Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes.
    
    (closes issue #16770)
     Reported by: jamicque
     Patches: 
           20100413__issue16770.diff.txt uploaded by tilghman (license 14)
           20100811__issue16770.diff.txt uploaded by tilghman (license 14)
     Tested by: jamicque
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  r282577 | dvossel | 2010-08-17 16:36:57 -0500 (Tue, 17 Aug 2010) | 16 lines
  
  Merged revisions 282576 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines
    
    fixes no default transport for temp peer creation in chan_sip
    
    (closes issue #17829)
    Reported by: falves11
    Patches:
          issue_17829.rev1.txt uploaded by russell (license 2)
          issue_17829.diff uploaded by dvossel (license 671)
    Tested by: falves11
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  r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 Aug 2010) | 6 lines
  
  ACCEPT message should respond with the new FORMAT2 ie
  
  (closes issue #17804)
  Reported by: tpanton
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  r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) | 4 lines
  
  fixes truncated uint64_t value in put_unaligned_uint64_t() function
  
  (issue #17804)
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  r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines
  
  Merged revisions 282467 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines
    
    Merged revisions 282430 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines
      
      Send a SRCCHANGE indication when we masquerade
      
      Masquerading a channel means that the src of the audio is potentially
      changing, so send a SRCCHANGE so that RTP-based media streams can get
      a new SSRC generated to reflect the change. Original patch by addix
      (along with lots of testing--thanks!).
      
      (closes issue #17007)
      Reported by: addix
      Patches: 
            1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006)
            srcchange.diff uploaded by twilson (license 396)
      Tested by: addix, twilson
      
      Review: https://reviewboard.asterisk.org/r/862/
    ........
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  r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
  
  Merged revisions 282469 via svnmerge from 
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  ........
    r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
    
    Add information about creating sounds files using
    the sounds tools publically available so that others can create their
    own sounds prompts using the same tools we use to generate sounds releases.
    This allows people creating their own prompts to sound consistent with
    the prompts available from the open source project.
    
    SWP-595
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See
http://glibc-bsd.alioth.debian.org/porting/PORTING

This patch gets Asterisk close to building on Debian kFreeBSD i386,
mainly by adding an extra test for __GLIBC__ in one or two (or more)
places.

OSARCH is set to 'kfreebsd-gnu'

DAHDI support (and support for chan_vpb) was not tested.

Review: https://reviewboard.asterisk.org/r/858/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282397 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines
  
  Fix our FRACKing issue with chan_iax2 a different way.
  
  Review: https://reviewboard.asterisk.org/r/861/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282367 65c4cc65-6c06-0410-ace0-fbb531ad65f3