3ca445e34c 
								
							
								 
							
						 
						
							
							
								
								Issue  #7541  - Handle multipart attachments to SIP messages - even if boundary is quoted.  
							
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								91e22d1df2 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 54373 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines
When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.
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								7bcf1d913a 
								
							
								 
							
						 
						
							
							
								
								Remove a couple of leftover debug messages  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								b100b69703 
								
							
								 
							
						 
						
							
							
								
								If we fail to create the SIP socket, then return -1 from reload_config() so  
							
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							that load_module() will return AST_MODULE_LOAD_DECLINE.  Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								f5590445cc 
								
							
								 
							
						 
						
							
							
								
								Fix VLDTMF reception  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								93fcd4a354 
								
							
								 
							
						 
						
							
							
								
								Change some text to properly state "On Hold", which was already done in trunk.  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								7ee02f585d 
								
							
								 
							
						 
						
							
							
								
								Merge team/russell/sla_rewrite  
							
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							This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4.  It is now functional and ready for testing.  However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample 
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
 - Add the ability to indicate HOLD state in NOTIFY messages.
 - Queue HOLD and UNHOLD control frames even if the channel is not bridged to
   another channel.
linkedlists.h:
 - Add support for rwlock based linked lists.
dial.c:
 - Add the ability to run ast_dial_start() without a reference channel to
   inherit information from.
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								ff17d402de 
								
							
								 
							
						 
						
							
							
								
								another dependency  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								2f6312a37e 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 53357 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines
Fix a few potential memory leaks with realtime users and peers. (issue #8999  reported by bsmithurst)
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								1c90803367 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 52843 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line
fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated)
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								e7a0e86756 
								
							
								 
							
						 
						
							
							
								
								Formatting  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								8e07358edf 
								
							
								 
							
						 
						
							
							
								
								Add some comments on queue system behaviour and how it affects the  
							
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							SIP channel
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								910898b7be 
								
							
								 
							
						 
						
							
							
								
								Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue  #8972  reported by mdu113)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53138  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								10ad9bb6d0 
								
							
								 
							
						 
						
							
							
								
								Remove quote from H.323 vendor string because due to compatibilities with CS1000 reported at www.voip-info.org  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53131  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								90a4b844a9 
								
							
								 
							
						 
						
							
							
								
								Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now  
							
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							considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								57fe6882ac 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 53103 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
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								09844a7f1a 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 53095 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines
Don't negotiate RFC2833 when not configured to do so. (issue #8799  reported by mdu113) 
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								97efd0be22 
								
							
								 
							
						 
						
							
							
								
								- Clean INC_COUNT flag when we decrement call counter  
							
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							- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
  before we destroy the dialog
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53085  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								6bb6bba6a3 
								
							
								 
							
						 
						
							
							
								
								Cleaning up the devicestate callback function  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53079  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								e86275c11c 
								
							
								 
							
						 
						
							
							
								
								Fix silly logic. We really want to write UDPTL frames out when the call is up.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53064  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								60ad44b8e9 
								
							
								 
							
						 
						
							
							
								
								we update the name on any first reply of our setup  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								6fd3418e40 
								
							
								 
							
						 
						
							
							
								
								chan_h323 is very stable, so let it built by default  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								9aab046002 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 53045 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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								29b7393d84 
								
							
								 
							
						 
						
							
							
								
								Only set the DTMF flag on the rtp structure if the DTMF mode is actually  
							
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							RFC2833, not just that it is not INFO.  This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936 )
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								362abb9deb 
								
							
								 
							
						 
						
							
							
								
								Drop the deprecated show commands since the original ones were changed back. (issue  #8937  reported by PCadach)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52856  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								a175571e39 
								
							
								 
							
						 
						
							
							
								
								Revert reprecation of h.323 gk cycle command from pre-1.4 version instead of duplicated h323 cycle gk  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								a46e6d74df 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 52762 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines
Fix the extraction of the timestamp from video frames.  It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795 , mihai)
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								6b462c95c9 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 52360 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines
Make the last context entry read in the dominant one. (issue #8918  reported by pj)
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								ed48c69f06 
								
							
								 
							
						 
						
							
							
								
								Drop out variables I accidentally put in.  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								6b08afd05d 
								
							
								 
							
						 
						
							
							
								
								Decrement onHold count if we are hung up on and still on hold. (issue  #8909  reported by alexh42)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52208  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								5ebd1ecf63 
								
							
								 
							
						 
						
							
							
								
								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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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52016  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								9352eda642 
								
							
								 
							
						 
						
							
							
								
								added fix from  #8899  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								6d2afb98b4 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 51966 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 Jan 2007) | 1 line
fixed the busy problem (dialstatus was not busy when we called a busy extension)
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								a207a31a97 
								
							
								 
							
						 
						
							
							
								
								Show capabilities *and* preference in general settings in "sip show settings"  
							
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							(reported by Clona/Telio - Thanks!)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51931  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								8f7ddbef0d 
								
							
								 
							
						 
						
							
							
								
								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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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51788  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								e65aeb27b5 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 50495,50506 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | 6 lines
* more additions to make the RESTART message work
* added fix for misdn_call to allow SETUPs with empty 
  extensions, replaced the strtok_r functions with strsep for that 
  (inspired by Sandro Cappellazzo, thanks)
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r50506 | crichter | 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line
when we get L2 UP, the L1 is UP definitely too, so we set the L1 state up as well.
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								19fdc9b14e 
								
							
								 
							
						 
						
							
							
								
								manually merged r49922 and r50335, because of conflicts. this commint includes addition of the ISDN RESTART Message  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								964d2d2e42 
								
							
								 
							
						 
						
							
							
								
								Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51615  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								5a3acb0511 
								
							
								 
							
						 
						
							
							
								
								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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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51558  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0e37d936da 
								
							
								 
							
						 
						
							
							
								
								Fix VLDTMF support in chan_gtalk.  AST_FRAME_DTMF and AST_FRAME_DTMF_END are  
							
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							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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51328  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								33235b40d6 
								
							
								 
							
						 
						
							
							
								
								Merge the changes from the /team/group/vldtmf_fixup branch.  
							
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							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...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								b62be97f7a 
								
							
								 
							
						 
						
							
							
								
								chan_zap compiles without libpri after committing 7877 patch  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								a4317ce870 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 51271 via svnmerge from  
							
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							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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							19 years ago  
				
					
						
							
							
								 
						
							
							
								1e3557c636 
								
							
								 
							
						 
						
							
							
								
								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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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51243  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								2b2f8d22f7 
								
							
								 
							
						 
						
							
							
								
								Instead of dividing the offset by 2 directly, make it more clear that the  
							
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							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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							19 years ago  
				
					
						
							
							
								 
						
							
							
								4244459e31 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 51197 via svnmerge from  
							
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							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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							19 years ago  
				
					
						
							
							
								 
						
							
							
								f986c4f2c3 
								
							
								 
							
						 
						
							
							
								
								Move rescheduling of lagrq/pings into the scheduler callback.  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								531d008066 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 51085 via svnmerge from  
							
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							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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							19 years ago  
				
					
						
							
							
								 
						
							
							
								915f9315e1 
								
							
								 
							
						 
						
							
							
								
								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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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50468  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								240ca25bea 
								
							
								 
							
						 
						
							
							
								
								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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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50466  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								9aca2b2a54 
								
							
								 
							
						 
						
							
							
								
								Fix chan_sip not working issue. Let's not prematurely return 0. (issue  #8783  reported by st41ker)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50377  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								1a33c38a15 
								
							
								 
							
						 
						
							
							
								
								- handle re-invites properly in sip_hangup()  
							
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							- Add some invitestate status changes just to be sure
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50124  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								3394598f93 
								
							
								 
							
						 
						
							
							
								
								Issue  #8677  - Handle failure of T.38 re-invite  
							
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							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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50006  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0f96f73768 
								
							
								 
							
						 
						
							
							
								
								Issue  #8524 , support multiple via header values (tardieu)  
							
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							Thanks!
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								be32fad9b8 
								
							
								 
							
						 
						
							
							
								
								We only need one forward declaration  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								484add6554 
								
							
								 
							
						 
						
							
							
								
								Issue 8735: Terminate state when extension is unavailable for subscription  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49925  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0a51642588 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 49889 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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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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							19 years ago  
				
					
						
							
							
								 
						
							
							
								dcbf36432e 
								
							
								 
							
						 
						
							
							
								
								Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								9803b9a357 
								
							
								 
							
						 
						
							
							
								
								Make codec_zap and chan_zap also depend on zaptel.  
							
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							This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49705  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								444adcb477 
								
							
								 
							
						 
						
							
							
								
								reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49676  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								fb010e49aa 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 49635 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines
ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly
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								ab4da71374 
								
							
								 
							
						 
						
							
							
								
								revert the dynamic_list insertion change... that was not the right thing to do  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49600  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								aed1467281 
								
							
								 
							
						 
						
							
							
								
								create the IAX2 processing threads as background threads so they will use smaller stacks  
							
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							when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49581  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								62b8b03294 
								
							
								 
							
						 
						
							
							
								
								It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49568  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								24bb7d865e 
								
							
								 
							
						 
						
							
							
								
								don't mark these allocations as 'cache' allocations when caching has been disabled  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49536  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								6b1817c0d1 
								
							
								 
							
						 
						
							
							
								
								if we're going to decrement the frame count when we free a frame, we should inrement it when we create one :-)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49523  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								497dd42772 
								
							
								 
							
						 
						
							
							
								
								only do IAX2 frame caching for voice and video frames  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49465  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								f7a47b38eb 
								
							
								 
							
						 
						
							
							
								
								don't define this type either if LOW_MEMORY is enabled  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49460  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								74258ab85e 
								
							
								 
							
						 
						
							
							
								
								don't do frame caching in LOW_MEMORY mode  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49458  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								fb52698667 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line
added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that.
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r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict.
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r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line
when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults.
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r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line
when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines
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r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line
added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. 
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r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines
* Added check for bridging in misdn_call to avoid setting echocancellation
  when 2 mISDN channels are involved and when bridging is set. That lead
  to a kernel panic before under different situations, because we switched 
  about 2 times between hardware bridging and echocancelation
* readded MISDN_URATE variable which got lost before, this should make app_v110
  work again
* fixed typo
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								40707f097a 
								
							
								 
							
						 
						
							
							
								
								Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49259  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								5edb7fa173 
								
							
								 
							
						 
						
							
							
								
								Small cleanup of add_t38sdp - it's always enabled at that point in the code  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49212  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								95758b4920 
								
							
								 
							
						 
						
							
							
								
								remove comment that is unrelated to this function  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49165  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								f195c5d622 
								
							
								 
							
						 
						
							
							
								
								check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49102  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								7db2ca152c 
								
							
								 
							
						 
						
							
							
								
								remove incomplete implementation of dnsmgr. Let's fix this in trunk.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49096  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								05716471b5 
								
							
								 
							
						 
						
							
							
								
								Use asprintf to build the channel names instead of custom function. I believe the custom function is doing some things that are not portable across all implementations. (issue  #8570  reported by hterag & issue  #8692  reported by nicolasg)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49070  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								37e5a39d76 
								
							
								 
							
						 
						
							
							
								
								Initialize the packet queue in load_module instead of just declaring the list with the default value. (issue  #8695  reported by ssokol)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49063  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								1e9c141c2d 
								
							
								 
							
						 
						
							
							
								
								Only include acl.h and lock.h once  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48997  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								4f157be79e 
								
							
								 
							
						 
						
							
							
								
								Only set rfc2833compensate flag once (handle_common_options)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48995  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								f1f8bbaabe 
								
							
								 
							
						 
						
							
							
								
								- Remove checking for T38 options twice. Keeping them in handle_common_options  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48993  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								0f5aae9688 
								
							
								 
							
						 
						
							
							
								
								make the option actually match the documentation  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48988  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								b2c8abbc6d 
								
							
								 
							
						 
						
							
							
								
								allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48987  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								d2b7e8b247 
								
							
								 
							
						 
						
							
							
								
								Be a bit more politically correct  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								bfe4bb0f1e 
								
							
								 
							
						 
						
							
							
								
								Issue  #8575  - Buggy cisco MWI support.  
							
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							Normally we try not to change our software for bugs in other devices. But in
this case, the Cisco phones are so widespread so we try to implement a fix while
waiting for a bugfix from Cisco.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								e25756dfac 
								
							
								 
							
						 
						
							
							
								
								- Make sure handle_common_options return 1 when we found a common option  
							
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							- Move uncommon (only global) option away from handle_common_options
Reported by rizzo. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48980  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								4ea530f2dc 
								
							
								 
							
						 
						
							
							
								
								Issue 8599 (rizzo) Change invitestate before re-sending invite with auth.  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48977  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								8a42650605 
								
							
								 
							
						 
						
							
							
								
								Fix bogus content-length in t38 sdp. (rizzo,  #8600 )  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48975  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								9cc04e026d 
								
							
								 
							
						 
						
							
							
								
								Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)  
							
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							git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48964  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								a1eee8d2d9 
								
							
								 
							
						 
						
							
							
								
								Fix a typo in an error message that indicated that the MGCP channel type could  
							
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							not be registered, instead of the correct type, OSS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48948  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
							
						 
						
							19 years ago  
				
					
						
							
							
								 
						
							
							
								53115ff688 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 48943 via svnmerge from  
							
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							https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | 3 lines
Check for the proper return value on an error in a call to mmap().
This was reported by Andy Wang on the asterisk-dev list.  Thanks!
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								1208869c00 
								
							
								 
							
						 
						
							
							
								
								Merged revisions 48939 via svnmerge from  
							
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r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines
Remove a couple of misplaced dots in log messages.  This was reported by
Andrea Spadaccini on the asterisk-dev mailing list.
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								66783cb46f 
								
							
								 
							
						 
						
							
							
								
								Note to self: Run make before committing...  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								ef2f05a518 
								
							
								 
							
						 
						
							
							
								
								Fix for issue 7774 - patch by alamantia  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								2703d88576 
								
							
								 
							
						 
						
							
							
								
								suppress compiler warnings in this module until it can be improved  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								11e61ea68a 
								
							
								 
							
						 
						
							
							
								
								Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue  #8604  reported by arkadia)  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								a0c21d30ac 
								
							
								 
							
						 
						
							
							
								
								Hold call structure lock in places where a qualify or peer action can destroy it.  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								195c66b4a1 
								
							
								 
							
						 
						
							
							
								
								Lock network retransmission queue in all places that it is used.  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								c1f1cf48ac 
								
							
								 
							
						 
						
							
							
								
								Issue  #8592  - treat 504 as 503 (imported from 1.2)  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								a8fe77b95b 
								
							
								 
							
						 
						
							
							
								
								Update to latest IANA spec  
							
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							19 years ago  
				
					
						
							
							
								 
						
							
							
								3706ecbbc5 
								
							
								 
							
						 
						
							
							
								
								Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted.  
							
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							19 years ago