Jason Parker 
							
						 
					 
					
						
						
							
						
						836c550ce3 
					 
					
						
						
							
							Merged revisions 81412 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
(closes issue #10621 )
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r81412 | qwell | 2007-08-31 13:44:44 -0500 (Fri, 31 Aug 2007) | 4 lines
Re-order dial options to be in line with the existing alpha order.
Issue 10621, initial patch by junky
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						2007-08-31 18:46:02 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						22114b509d 
					 
					
						
						
							
							Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-08-08 21:44:58 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						9ef1b0a974 
					 
					
						
						
							
							Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78278  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-08-06 21:52:30 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						4e0947c5f1 
					 
					
						
						
							
							Convert code that checks the _softhangup member of ast_channel directory to use  
						
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						the ast_check_hangup() funciton.  This function takes scheduled hangups into
account.
(closes issue #10230 , patch by Juggie)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77858  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-08-01 15:39:54 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						ceca4d97e1 
					 
					
						
						
							
							These fixes take care of two problems: a complaint in asterisk-dev that goto's aren't working in trunk, a side effect of the move to commas as arg seps in apps and funcs; and a problem I spotted myself with dial's 'e' option, where gotos were off by one, because I forgot to set the AUTOLOOP flag in the peer channel.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77520  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-27 15:46:20 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						f8483a0d04 
					 
					
						
						
							
							Do a massive conversion for using the ast_verb() macro  
						
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						(closes issue #10277 , patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
   ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-26 15:49:18 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						55b1ee298e 
					 
					
						
						
							
							Merge the dialplan_aesthetics branch.  Most of this patch simply converts applications  
						
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						using old methods of parsing arguments to using the standard macros.  However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar).  Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-23 19:51:41 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						0e969271ae 
					 
					
						
						
							
							After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-19 23:24:27 +00:00 
						 
				 
			
				
					
						
							
							
								Mark Michelson 
							
						 
					 
					
						
						
							
						
						ee6d59eef2 
					 
					
						
						
							
							Merged revisions 75405 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul 2007) | 6 lines
Fixing an error I made earlier. ast_fileexists can return -1 on failure, so I need to be sure that we only enter the if
statement if it is successful.
Related to my fix to issue #10186 
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75406  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2007-07-17 20:05:19 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						8a7732f067 
					 
					
						
						
							
							via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-17 19:40:29 +00:00 
						 
				 
			
				
					
						
							
							
								Jason Parker 
							
						 
					 
					
						
						
							
						
						766121a5bc 
					 
					
						
						
							
							Fix an incorrect parenthesization (TODO: Find a better word) in app_dial  
						
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						Pointed out by Fanzhou Zhao
Closes issue #10216 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75351  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-17 12:01:05 +00:00 
						 
				 
			
				
					
						
							
							
								Mark Michelson 
							
						 
					 
					
						
						
							
						
						ce8f95d750 
					 
					
						
						
							
							Merged revisions 75253 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul 2007) | 8 lines
Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified.
This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up).
If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will
still continue. 
(closes issue #10186 , reported by jon, patched by me)
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						2007-07-16 18:18:19 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						b8cd949cce 
					 
					
						
						
							
							Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-16 14:39:29 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						96a646734f 
					 
					
						
						
							
							It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-16 13:35:20 +00:00 
						 
				 
			
				
					
						
							
							
								Olle Johansson 
							
						 
					 
					
						
						
							
						
						a1b9cbcd31 
					 
					
						
						
							
							Implementation of a feature that will disable "missed calls" counters on SIP phones.  
						
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						If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-07-09 08:27:37 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						8b93f50dfc 
					 
					
						
						
							
							Merged revisions 73053 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r73053 | tilghman | 2007-07-03 07:38:53 -0500 (Tue, 03 Jul 2007) | 10 lines
Merged revisions 73052 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) | 2 lines
RetryDial should accept a 0 argument, but it does not, because atoi does not distinguish between 0 and error (closes issue #10106 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73054  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2007-07-03 12:40:26 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						a1bc823136 
					 
					
						
						
							
							Issue 9990 - New API ast_mkdir, which creates parent directories as necessary (and is faster than an outcall to mkdir -p)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71040  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-22 04:35:12 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						2462d5ab4f 
					 
					
						
						
							
							Cleaning up a small disaster I created earlier  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70607  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-20 23:26:07 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						57526b35cc 
					 
					
						
						
							
							As per 9228, now app_queue should have the proper machinery to do gosubs.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70531  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-20 21:38:49 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						ce2c52d519 
					 
					
						
						
							
							Merged revisions 70445 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r70445 | tilghman | 2007-06-20 14:29:23 -0500 (Wed, 20 Jun 2007) | 10 lines
Merged revisions 70444 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) | 2 lines
Issue 9997 - Timelimit times out the wrong channel
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						2007-06-20 19:30:31 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						704c756c4a 
					 
					
						
						
							
							Merge work to make U(...) option work for Dial  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70358  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-20 17:35:08 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						866bbaa515 
					 
					
						
						
							
							Via bug9228, no way to create macros via AEL, and some of the apps allow you to call macros..., I modded the apps that allow macro calls to allow gosubs calls also, to make them AEL compliant.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70161  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-19 23:36:34 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						055d82cbce 
					 
					
						
						
							
							Add a massive set of changes for converting to use the ast_debug() macro.  
						
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						(issue #9957 , patches from mvanbaak, caio1982, critch, and dimas)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-14 19:39:12 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						9e0458e9f1 
					 
					
						
						
							
							Completely remove all of the code related to jumping to priority n + 101.  yay!  
						
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						(issue #9926 , caio1982)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68970  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-12 15:58:28 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						2492bf26fb 
					 
					
						
						
							
							Merged revisions 68071 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r68071 | file | 2007-06-07 10:21:59 -0400 (Thu, 07 Jun 2007) | 10 lines
Merged revisions 68070 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 lines
Allow the 'g' option to work if used with the 'S' option. (issue #9888  reported by gasparz)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68072  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2007-06-07 14:23:21 +00:00 
						 
				 
			
				
					
						
							
							
								Tilghman Lesher 
							
						 
					 
					
						
						
							
						
						9d05ff8ed5 
					 
					
						
						
							
							Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-06-06 21:20:11 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						bae04fd90e 
					 
					
						
						
							
							Merged revisions 67066 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2 lines
Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661  reported by dimas)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67067  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2007-06-04 18:00:24 +00:00 
						 
				 
			
				
					
						
							
							
								Steve Murphy 
							
						 
					 
					
						
						
							
						
						4572edae31 
					 
					
						
						
							
							Merged revisions 65200 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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						2007-05-18 22:33:51 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						83d0b0417c 
					 
					
						
						
							
							Merged revisions 64756 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) | 3 lines
Increase the size of a buffer to support longer dial strings for channels.
(issue #9291 , reported and fix suggested by meni)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64757  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2007-05-17 16:49:50 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						e2871220e2 
					 
					
						
						
							
							Merged revisions 61656 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r61656 | file | 2007-04-13 15:17:08 -0400 (Fri, 13 Apr 2007) | 10 lines
Merged revisions 61655 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 lines
Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140)
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						2007-04-13 19:18:46 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						be874b92d3 
					 
					
						
						
							
							Remove unused instances of unnamed enums.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61116  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-04-09 22:49:32 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						840f1e61e0 
					 
					
						
						
							
							Merged revisions 60798 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r60798 | file | 2007-04-08 21:03:14 -0400 (Sun, 08 Apr 2007) | 10 lines
Merged revisions 60797 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 lines
When calling a device that then forwards us elsewhere... we have to make our channels compatible if it is the only channel being dialed. (issue #9445  reported by marcelbarbulescu)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60799  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2007-04-09 01:06:56 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						726928cb60 
					 
					
						
						
							
							Properly hangup the original dialed channel, not the new channel that appeared from the forwarding. (issue  #9161  reported by PhilSmith)  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57011  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2007-02-27 22:17:42 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						fb9e48ca9e 
					 
					
						
						
							
							Merged revisions 55154 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r55154 | file | 2007-02-16 22:55:30 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 55153 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2 
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r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Answer the channel before recording privacy information. (issue #8926  reported by lmamane)
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						2007-02-17 03:57:23 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						8e92c73a1d 
					 
					
						
						
							
							Merged revisions 54924 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r54924 | file | 2007-02-16 13:51:15 -0500 (Fri, 16 Feb 2007) | 2 lines
Need to check macro extension as well as macro context for directed pickup.
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						2007-02-16 18:53:17 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						821c941976 
					 
					
						
						
							
							Merged revisions 54884 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r54884 | file | 2007-02-16 12:02:35 -0500 (Fri, 16 Feb 2007) | 2 lines
Allow directed pickup to pick up the real context instead of the macro context if a Macro is used. (issue #8984  reported by jamesb63)
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						2007-02-16 17:07:10 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						a11b56a8e5 
					 
					
						
						
							
							Merged revisions 54623 via svnmerge from  
						
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r54623 | file | 2007-02-15 11:19:39 -0500 (Thu, 15 Feb 2007) | 10 lines
Merged revisions 54622 via svnmerge from 
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r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines
Use a separate variable to indicate execution should continue instead of the return value. (issue #8842  reported by pluto70)
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						2007-02-15 16:24:13 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						68a66656e6 
					 
					
						
						
							
							Merged revisions 54481 via svnmerge from  
						
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r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2 lines
Forward begin DTMF frames as well as end. (issue #9068  reported by mhardeman)
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						2007-02-14 21:10:53 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						32cd307d6f 
					 
					
						
						
							
							Merged revisions 53749 via svnmerge from  
						
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r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2 lines
Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969  reported by christianbee)
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						2007-02-09 19:39:26 +00:00 
						 
				 
			
				
					
						
							
							
								Russell Bryant 
							
						 
					 
					
						
						
							
						
						ce321f87e9 
					 
					
						
						
							
							Merged revisions 53136 via svnmerge from  
						
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r53136 | russell | 2007-02-03 14:44:20 -0600 (Sat, 03 Feb 2007) | 12 lines
Merged revisions 53133 via svnmerge from 
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r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines
set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application
exits early because of invalid arguments instead of just leaving it empty.
(issue #8975 )
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						2007-02-03 20:46:36 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						a378353ecc 
					 
					
						
						
							
							Merged revisions 50298 via svnmerge from  
						
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r50298 | file | 2007-01-09 23:55:13 -0500 (Tue, 09 Jan 2007) | 10 lines
Merged revisions 50295 via svnmerge from 
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r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 lines
Add another return value to dial_exec_full that indicates execution is going to continuing at a new extension/context/priority and to just let it slide. (issue #8598  reported by jon)
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						2007-01-10 04:56:48 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						83f52ed5e2 
					 
					
						
						
							
							better name for struct dial_localuser.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48581  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2006-12-19 16:36:45 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						a755ec928a 
					 
					
						
						
							
							introduce a temporary variable for tmp->chan to shorten expressions.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48575  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2006-12-19 09:58:40 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						9c81431c93 
					 
					
						
						
							
							stop what i think is a memory leak in case Dial fails to  
						
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						connect to a channel.
Before committing to 1.4 i would like some other people to
review and test this fix - thanks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48574  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2006-12-19 09:33:57 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						9d4531d636 
					 
					
						
						
							
							move a large block related to privacy handling to a separate function.  
						
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						git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48573  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2006-12-19 09:15:23 +00:00 
						 
				 
			
				
					
						
							
							
								Kevin P. Fleming 
							
						 
					 
					
						
						
							
						
						359a553961 
					 
					
						
						
							
							Merged revisions 48193 via svnmerge from  
						
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r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48192 via svnmerge from 
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
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						2006-12-01 23:39:59 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						bbba4e5a20 
					 
					
						
						
							
							better fix for the previous bug.  
						
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						In general this code needs a deep revision, because the body of
do_forward() deletes/overwrites the output channel without freeing
the resouce in some cases, and without notifying the caller.
Also, on FreeBSD with MALLOC_OPTIONS set i am seeing various panics
(duplicate freee etc.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47881  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2006-11-21 11:53:06 +00:00 
						 
				 
			
				
					
						
							
							
								Luigi Rizzo 
							
						 
					 
					
						
						
							
						
						64a9c28c3b 
					 
					
						
						
							
							do not ast_hangup() on a NULL channel.  
						
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						In the original code this would happen in the case of
	o->forwards >=  AST_MAX_FORWARDS
Likely an 1.2/1.4 isse as well - please someone have a look,
while I am hunting a few more similar panics now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47880  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2006-11-21 11:07:30 +00:00 
						 
				 
			
				
					
						
							
							
								Joshua Colp 
							
						 
					 
					
						
						
							
						
						af51be05a6 
					 
					
						
						
							
							Merged revisions 47850 via svnmerge from  
						
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						https://origsvn.digium.com/svn/asterisk/branches/1.4 
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r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2 lines
Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382  reported by jiddings)
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						2006-11-20 15:55:58 +00:00 
						 
				 
			
				
					
						
							
							
								Jason Parker 
							
						 
					 
					
						
						
							
						
						938c4bdc29 
					 
					
						
						
							
							Merged revisions 47782 via svnmerge from  
						
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r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2 lines
Fix a couple of typos.  Initially pointed out by mrobinson.
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						2006-11-16 23:20:52 +00:00