Joshua Colp 
							
						 
					 
					
						
						
							
						
						300f980223 
					 
					
						
						
							
							Use atomic operation functions for use/ringing/hold manipulation.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52240  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-25 18:10:18 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						b8d6cbcd3f 
					 
					
						
						
							
							Merged revisions 52210 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 lines
Drop out variables I accidentally put in.
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						2007-01-25 17:51:35 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						afb9151e19 
					 
					
						
						
							
							Merged revisions 52208 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 lines
Decrement onHold count if we are hung up on and still on hold. (issue #8909  reported by alexh42)
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						2007-01-25 17:17:56 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						2396e24e65 
					 
					
						
						
							
							Merged revisions 52016 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 lines
Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535  reported by ivoc)
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						2007-01-24 18:04:47 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						273f1d78c7 
					 
					
						
						
							
							Merged revisions 51931 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 lines
Show capabilities *and* preference in general settings in "sip show settings"
(reported by Clona/Telio - Thanks!)
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						2007-01-24 09:42:31 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						ee3ab150f6 
					 
					
						
						
							
							Merged revisions 51788 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897  reported by junky)
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						2007-01-23 22:59:55 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						ef4db783c6 
					 
					
						
						
							
							Issue  #8817  - Registry corruption when packet retransmits fail. (tootai, patchy by oej)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51659  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-23 15:36:01 +00:00 
						 
				 
			
				
					
						
							
							
								Paul Cadach 
							
						 
					 
					
						
						
							
						
						cd61f69a83 
					 
					
						
						
							
							Merged revisions 51615 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) | 1 line
Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen)
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						2007-01-23 06:56:26 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						34df128519 
					 
					
						
						
							
							Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue  #7812  reported by drorlb)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51560  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-23 03:15:04 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						5a25c156c6 
					 
					
						
						
							
							Merged revisions 51558 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 lines
Only change audio formats on the channel if we have an audio format to change to. (issue #8535  reported by ivoc)
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						2007-01-23 03:02:09 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						1a5dfca2a1 
					 
					
						
						
							
							Remove (to quote Rizzo) "useless" variable.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51405  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-22 19:00:25 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						3275357f20 
					 
					
						
						
							
							Merged revisions 51328 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk.  AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing.  So, a digit would have been interpreted incorrectly
here.  Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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						2007-01-19 19:09:04 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						dcca8f345f 
					 
					
						
						
							
							Merged revisions 51311 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged.  So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio.  However,
since there was no audio coming in, the DTMF_END was never generated.  This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf).  If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback.  This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597 , maybe others...)
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						2007-01-19 18:06:03 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						a52fa7e5ad 
					 
					
						
						
							
							sizeof() is compatible with format %d so don't be too  
						
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						picky on printf formats.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51309  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-19 17:45:46 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						455ea1c99e 
					 
					
						
						
							
							remove variable declaration in the middle of a block  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51308  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-19 17:38:28 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						0a8d59d88a 
					 
					
						
						
							
							whoops, fix a cut&paste error...  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51298  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-19 16:35:37 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						03fc21712a 
					 
					
						
						
							
							slight change to the initialization of a structure,  
						
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						also using '\0' to make it clear we need a (char)0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51297  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-19 16:32:05 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						3fefb9d516 
					 
					
						
						
							
							include "asterisk/zapata.h" to get the zaptel headers.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51295  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-19 16:28:24 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						dc31eaa025 
					 
					
						
						
							
							include "asterisk/zapata.h" instead of looking  
						
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						directly for the zaptel.h and tonezone.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51292  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-19 16:16:19 +00:00 
						 
				 
			
				
					
						
							
							
								Dwayne M. Hubbard 
							
						 
					 
					
						
						
							
						
						18c23101a5 
					 
					
						
						
							
							Merged revisions 51274 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007) | 3 lines
chan_zap compiles without libpri after committing 7877 patch
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						2007-01-19 00:28:25 +00:00 
						 
				 
			
				
					
						
							
							
								Dwayne M. Hubbard 
							
						 
					 
					
						
						
							
						
						33243c6624 
					 
					
						
						
							
							Merged revisions 51272 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51272 | dhubbard | 2007-01-18 17:56:49 -0600 (Thu, 18 Jan 2007) | 11 lines
Merged revisions 51271 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads.  Reset configuration variables to default values prior to parsing configuration file.
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						2007-01-19 00:06:35 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						1a06a58250 
					 
					
						
						
							
							Merged revisions 51243 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 lines
Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840  reported by mdu113)
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						2007-01-18 18:39:21 +00:00 
						 
				 
			
				
					
						
							
							
								Paul Cadach 
							
						 
					 
					
						
						
							
						
						ae286e1843 
					 
					
						
						
							
							Update ast_append_ha() usage  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51221  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-18 06:59:22 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						45bd0c4f3f 
					 
					
						
						
							
							Merged revisions 51204 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | 4 lines
Instead of dividing the offset by 2 directly, make it more clear that the
offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)
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						2007-01-17 23:31:40 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						c56f9184e8 
					 
					
						
						
							
							Merged revisions 51198 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51198 | russell | 2007-01-17 15:18:35 -0600 (Wed, 17 Jan 2007) | 11 lines
Merged revisions 51197 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines
Move the check for a failure of ast_channel_alloc() to before locking the
pvt structure again.  Otherwise, on a failure, this will cause a deadlock.
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						2007-01-17 21:20:22 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						31224a6b52 
					 
					
						
						
							
							Merged revisions 51172 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2 lines
Move rescheduling of lagrq/pings into the scheduler callback.
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						2007-01-17 00:50:59 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						04426fab2c 
					 
					
						
						
							
							Add support for G729 passthrough with Sigma Designs boards. (issue  #8829  reported by ywalther)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51144  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-16 17:23:31 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						3d91c0a0c0 
					 
					
						
						
							
							IAX2 remote variables - Bug 7619  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51123  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-16 08:38:59 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						bf5c607c69 
					 
					
						
						
							
							Merged revisions 51087 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r51087 | file | 2007-01-16 00:55:23 -0500 (Tue, 16 Jan 2007) | 10 lines
Merged revisions 51085 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines
Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296  reported by gkloepfer)
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						2007-01-16 05:56:42 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						b3288f79fe 
					 
					
						
						
							
							Feature:  allow soundcard to be used in both modes (autoanswer and not),  
						
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						selectable by how it is called in the dialplan.  This allows a speaker
system hooked up to the soundcard to be used for both ring notification,
as well as paging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50847  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-15 02:28:32 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						91dfb27494 
					 
					
						
						
							
							Get rid of unneeded code, fix a spelling mistake, and use registry state a bit more. (issue  #8402  reported by rizzo)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50709  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-13 04:56:25 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						033d849bda 
					 
					
						
						
							
							Drop trunkrealloc option and just have the maximum size be a configurable option. This is per Kevin's comments on -dev and my own thoughts after I put the previous option in.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50698  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-13 04:04:04 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						0d781f9553 
					 
					
						
						
							
							Ensure error variable is set to 0 or else we might get false error messages. (issue  #8798  reported by tootai, fix by anthonyl)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50677  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-13 03:37:10 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						c4b4615dcd 
					 
					
						
						
							
							Merge in trunkrealloc option for chan_iax2. (issue  #8267  reported by marcodmb, branch by anthonyl)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50676  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-01-13 03:26:04 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						469cf8ab2a 
					 
					
						
						
							
							Merged revisions 50468 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 lines
Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing.
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						2007-01-11 05:54:14 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						4942fd94d2 
					 
					
						
						
							
							Merged revisions 50466 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655  reported by mnicholson)
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						2007-01-11 05:21:03 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						47c515768c 
					 
					
						
						
							
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r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2 lines
Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783  reported by st41ker)
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						2007-01-10 18:34:45 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						f215636ef3 
					 
					
						
						
							
							Allow usedistinctiveringdetection and distinctiveringaftercid to be reset during a reload. (issue  #8739  reported by tzafrir)  
						
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						2007-01-10 04:21:46 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						444acad95b 
					 
					
						
						
							
							Covert some spaces to tabs, and put a list of defines in an enum.  
						
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						2007-01-10 02:50:06 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						3246ad09a4 
					 
					
						
						
							
							Based on the following patch, changed for trunk...  
						
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r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 lines
- handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure
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						2007-01-09 12:25:33 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						5d4f272a90 
					 
					
						
						
							
							Merged revisions 50006 via svnmerge from  
						
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r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines
Issue #8677  - Handle failure of T.38 re-invite
This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).
To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.
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						2007-01-08 14:31:16 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						835e2ed684 
					 
					
						
						
							
							Merged revisions 49983 via svnmerge from  
						
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r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3 lines
Issue #8524 , support multiple via header values (tardieu)
	Thanks!
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						2007-01-08 13:34:56 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						975dba04bc 
					 
					
						
						
							
							Merged revisions 49945 via svnmerge from  
						
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r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2 lines
We only need one forward declaration
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						2007-01-08 09:22:53 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						7158f48a79 
					 
					
						
						
							
							Merged revisions 49925 via svnmerge from  
						
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r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2 lines
Issue 8735: Terminate state when extension is unavailable for subscription
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						2007-01-08 08:57:35 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						585e0e3e09 
					 
					
						
						
							
							Merged revisions 49890 via svnmerge from  
						
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r49890 | file | 2007-01-08 00:11:54 -0500 (Mon, 08 Jan 2007) | 10 lines
Merged revisions 49889 via svnmerge from 
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r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 lines
Ensure we use the default refresh value of 60 if the remote server does not send one. (issue #8746  reported by maethor)
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						2007-01-08 05:13:24 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						4866a44535 
					 
					
						
						
							
							Merged revisions 49831 via svnmerge from  
						
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r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007) | 2 lines
Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)
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						2007-01-07 21:26:43 +00:00 
						 
				 
			
				
					
						
							
							
								Kevin P. Fleming 
							
						 
					 
					
						
						
							
						
						37182c873e 
					 
					
						
						
							
							finish const-ifying ast_func_read()  
						
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						2007-01-06 00:13:33 +00:00 
						 
				 
			
				
					
						
							
							
								Kevin P. Fleming 
							
						 
					 
					
						
						
							
						
						cd73a483f1 
					 
					
						
						
							
							const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here  
						
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						2007-01-05 23:32:42 +00:00 
						 
				 
			
				
					
						
							
							
								Jason Parker 
							
						 
					 
					
						
						
							
						
						cdd63e685d 
					 
					
						
						
							
							Merged revisions 49705 via svnmerge from  
						
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r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4 lines
Make codec_zap and chan_zap also depend on zaptel.
This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel.
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						2007-01-05 23:17:45 +00:00 
						 
				 
			
				
					
						
							
							
								Kevin P. Fleming 
							
						 
					 
					
						
						
							
						
						87b9abc892 
					 
					
						
						
							
							Merged revisions 49676 via svnmerge from  
						
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r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) | 2 lines
reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases
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						2007-01-05 22:43:18 +00:00