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2009-10-20Merged revisions 224774 via svnmerge from file1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224775 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-20Merged revisions 213327 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 | mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7 lines Fix a crash by checking the proper pointer for validity before deferencing it. (closes issue #15751) Reported by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license 242) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@213346 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-9/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@211551 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Merged revisions 209098 via svnmerge from dbrooks1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@209221 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-25Merged revisions 203376 via svnmerge from russell1-1/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines Merged revisions 203375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203377 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-19Merged revisions 201829 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) | 13 lines Merged revisions 201828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@201830 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-03Merged revisions 198856 via svnmerge from dvossel1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198889 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-20Merged revisions 195698 via svnmerge from file1-5/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195702 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-06Merged revisions 192861 via svnmerge from jpeeler1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) | 17 lines Merged revisions 192858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines Make ParkedCall application stop execution of the dialplan after hang up Just changed park_exec to always return non-zero. I really wasn't entirely sure at first if this was a bug. Decided it was since it would be surprising when not using ParkedCall in the dialplan to hang up and have dialplan execution continue. (closes issue #14555) Reported by: francesco_r ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@192865 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-05Merged revisions 192462 via svnmerge from file1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) | 15 lines Merged revisions 192454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines Fix an incorrect assumption that certain values on the channel will always exist when they may not. The CDR code involved with bridges wrongly assumed that the currently executing application and data values will always exist. It is possible for this to be false when call forwarding is involved. (closes issue #14984) Reported by: gincantalupo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@192467 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-27Fix announcement of parking space after transfer to park.jpeeler1-1/+1
Don't set the orig_chan_name in masq_park_call when the announcement is to be played. This change was copied from all the other working 1.6 branches. (closes issue #14863) Reported by: noahisaac git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@190829 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-22Merged revisions 189951 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines Fix call parking callback. Pipes -> Commas. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@189984 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183436 via svnmerge from dvossel1-6/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@183437 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183172 via svnmerge from dvossel1-49/+85
https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@183199 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Merged revisions 182525 via svnmerge from kpfleming1-2/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182526 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13Merged revisions 182029 via svnmerge from mmichelson1-6/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182064 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179841 via svnmerge from file1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179842 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 178986 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178987 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178828 via svnmerge from murf1-2/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178866 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-19Merged revisions 177356 via svnmerge from jpeeler1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177357 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177226 via svnmerge from dvossel1-30/+45
https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177227 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176708 via svnmerge from jpeeler1-0/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176710 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Fix mistake in merging conflict from 175299.jpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175298 via svnmerge from jpeeler1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175299 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175188 via svnmerge from jpeeler1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175189 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173500 via svnmerge from jpeeler1-38/+65
https://origsvn.digium.com/svn/asterisk/trunk ................ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173546 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Merged revisions 173067 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173068 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Merged revisions 172580 via svnmerge from twilson1-15/+265
https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172635 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 172063 via svnmerge from murf1-12/+36
https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172065 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-20Merged revisions 169510 via svnmerge from twilson1-2/+24
https://origsvn.digium.com/svn/asterisk/trunk ........ r169510 | twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines Make a proper builtin attended transfer to parking work This is an ugly hack from 1.4 that allows the timeout callback from a parked call to use the right channel name for the callback when the park is done with a builtin attended transfer (that isn't completed early). This hasn't ever worked in trunk and no one has complained yet, so eh. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@169541 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-20Merged revisions 169486 via svnmerge from twilson1-11/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009) | 13 lines Merged revisions 169485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) | 6 lines Don't play audio to the channel if we've masqueraded (closes issue #14066) Reported by: bluefox Tested by: otherwiseguy, bluefox ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@169487 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-16Merged revisions 168941 via svnmerge from twilson1-5/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009) | 19 lines Merged revisions 168716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines Convert call to park_call_full to masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded parking, otherwise we will try to call ast_hangup() in __pbx_run() and in do_parking_thread() and then promptly crash. (closes issue #14215) Reported by: waverly360 Tested by: otherwiseguy (closes issue #14228) Reported by: kobaz Tested by: otherwiseguy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@168980 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-24Merged revisions 166665 via svnmerge from murf1-83/+90
https://origsvn.digium.com/svn/asterisk/trunk Due to non-symmetrical updating, I had some fairly interesting conflicts to straighten out in this release. The changes were such that I was compelled to run thru all the same tests as trunk, which turned up some problems, which I fixed. ................ r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@166729 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164203 via svnmerge from russell1-1/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@164279 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163094 via svnmerge from russell1-33/+39
https://origsvn.digium.com/svn/asterisk/trunk ................ r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008) | 19 lines Merged revisions 163092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines Fix an issue that made it so you could only have a single caller executing a custom feature at a time. This was especially problematic when custom features ran for any appreciable amount of time. The fix turned out to be quite simple. The dynamic features are now stored in a read/write list instead of a list using a mutex. (closes issue #13478) Reported by: neutrino88 Fix suggested by file ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@163095 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-02Merged revisions ↵tilghman1-4/+3
147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ ................ r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ ................ r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ ................ r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ ................ r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ ................ r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ ................ r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines Some compilers warn, some don't. Fixing. ........ ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ ................ r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ ................ r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ ................ r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@160387 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-29Merged revisions 159818 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@159855 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-21Merged revisions 158484 via svnmerge from murf1-13/+18
https://origsvn.digium.com/svn/asterisk/trunk ................ r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | 19 lines Merged revisions 158483 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines (closes issue #13871) Reported by: mdu113 This one is totally my fault. The code doesn't even create a bridge CDR if the channel CDR has POST_DISABLED. I didn't check for that at the end of the bridge. Fixed with a few small insertions. Tested. Looks good. No cdr generated, no crash, no unnecc. data objects created either. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@158545 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-18Merged revisions 157306 via svnmerge from mmichelson1-0/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@157307 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-09Merged revisions 155554 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@155555 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31Merged revisions 153181 via svnmerge from twilson1-1/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@153265 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-01Merged revisions 145606 via svnmerge from mmichelson1-3/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r145606 | mmichelson | 2008-10-01 17:23:50 -0500 (Wed, 01 Oct 2008) | 11 lines Okay, this should really do it now. While I did manage to fix blind transfers with my last commit here, I also caused an unwanted side-effect. That is, only the first priority of the 'h' extension would be executed when a blind transfer occurred instead of all priorities. Essentially, my last commit corrected the return value of ast_bridge_call. However, the implementation still was not 100% correct. Now it is. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@145607 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-01Merged revisions 145579 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r145579 | mmichelson | 2008-10-01 16:33:11 -0500 (Wed, 01 Oct 2008) | 4 lines if (!(x) == 0) is the same as if (x). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@145581 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-01Merged revisions 145553 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r145553 | mmichelson | 2008-10-01 16:06:26 -0500 (Wed, 01 Oct 2008) | 13 lines The logic surrounding the return value of ast_spawn_extension within ast_bridge_call was reversed. This problem was observed when a blind transfer placed from the callee channel of a test call failed. While the problem I am solving here is exactly the same as what was reported in issue #13584, the difference is that this fix I am applying is trunk-only. Issue #13584 was reported against the 1.4 branch, and my tests of 1.4's blind transfers appear to work fine. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@145556 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-23Merged revisions 144067 via svnmerge from murf1-43/+76
https://origsvn.digium.com/svn/asterisk/trunk ................ r144067 | murf | 2008-09-23 10:52:32 -0600 (Tue, 23 Sep 2008) | 37 lines Merged revisions 144066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | 29 lines (closes issue #13489) Reported by: DougUDI Tested by: murf (closes issue #13490) Reported by: seanbright Tested by: murf (closes issue #13467) Reported by: edantie Tested by: murf, edantie, DougUDI This crash happens because we are unsafely handling old pointers. The channel whose cdr is being handled, has been hung up and destroyed already. I reorganized the code a bit, and tried not to lose the fork-cdr-chain concepts of the previous code. I now verify that the 'previous' channel (the channel we had when the bridge was started), still exists, by looking it up by name in the channel list. I also do not try to reset the CDR's of channels involved in bridges. Testing shows it solves the crash problem, and should not negatively impact previous fixes involving CDR's generated during/after blind transfers. (The reason we need to reset the CDR's on the "beginning" channels in the first place). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@144068 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142676 via svnmerge from murf1-0/+44
https://origsvn.digium.com/svn/asterisk/trunk ................ r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@142677 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-11Merged revisions 142576 via svnmerge from murf1-24/+42
https://origsvn.digium.com/svn/asterisk/trunk ................ r142576 | murf | 2008-09-11 17:12:53 -0600 (Thu, 11 Sep 2008) | 28 lines Merged revisions 142575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | 20 lines (closes issue #13364) Reported by: mdu113 Well, fundamentally, the problems revealed in 13364 are because of the ForkCDR call that is done before the dial. When the bridge is in place, it's dealing with the first (and wrong) cdr in the list. So, I wrote a little func to zip down to the first non-locked cdr in the chain, and thru-out the ast_bridge_call, these results are used instead of raw chan->cdr and peer->cdr pointers. This shouldn't affect anyone who isn't forking cdrs before a dial, and should correct the cdr's of those that do. So, this change ends up correcting the dstchannel and userfield; the disposition was fixed by a previous patch, it was OK coming into this problem. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@142631 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-10Merged revisions 142475 via svnmerge from murf1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r142475 | murf | 2008-09-10 16:11:27 -0600 (Wed, 10 Sep 2008) | 38 lines Merged revisions 142474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | 30 lines (closes issue #12318) Reported by: krtorio I made a small change to the code that handles local channel situations. In that code, I copy the answer time from the peer cdr, to the bridge_cdr, but I wasn't also copying the disposition from the peer cdr. So, Now I copy the disposition, and I've tested against these cases: 1. phone 1 never answers the phone; no cdr is generated at all. this should show up as a manager command failure or something. 2. phone 2 never answers. CDR is generated, says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4. phone 2 answers: CDR is generated, times are correct; disposition is ANSWERED, which is correct. The start time is the time that the manager dialed the first phone. The answer time is the time the second phone picks up. I purposely left the cid and src fields blank; since this call really originates from the manager, there is no 'easy' data to put in these fields. If you feel strongly that these fields should be filled in, re-open this bug and I'll dig further. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@142477 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-09Merged revisions 142064 via svnmerge from russell1-0/+17
https://origsvn.digium.com/svn/asterisk/trunk ................ r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) | 13 lines Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@142065 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-04Merged revisions 141039 via svnmerge from jpeeler1-4/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines (closes issue #11979) Fixes multiple parking problems: Crash when executing a park on an extension dialed by AGI due to not returning the proper return code. Crash when using a builtin feature that was a subset of a enabled dynamic feature. Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@141086 f38db490-d61c-443f-a65b-d21fe96a405b
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