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2008-09-03Merged revisions 140749 via svnmerge from murf1-11/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | 11 lines Merged revisions 140747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug. For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@140827 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-03Merged revisions 140692 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@140826 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-03Merged revisions 140691 via svnmerge from murf2-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@140825 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-03Merged revisions 140817 via svnmerge from russell1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) | 12 lines Merged revisions 140816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) | 4 lines Don't freak out if the poll emulation receives NULL for the pollfds array (closes issue #13307) Reported by: jcovert ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@140818 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-25Merged revisions 139770 via svnmerge from murf2-47/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines This patch reverts the changes made via 139347, and 139635, as users are seeing adverse difference. I will un-close 13251. Back to the drawing board/ concept/ beginning/ whatever! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@139774 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-22Merged revisions 139662 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | 14 lines Merged revisions 139635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines I found some problems with the code I committed earlier, when I merged them into trunk, so I'm coming back to clean up. And, in the process, I found an error in the code I added to trunk and 1.6.x, that I'll fix using this patch also. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@139671 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-22Merged revisions 139627 via svnmerge from murf2-0/+47
https://origsvn.digium.com/svn/asterisk/trunk ................ r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@139628 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-20Merged revisions 139083 via svnmerge from murf1-4/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | 20 lines Merged revisions 139074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines (closes issue #13263) Reported by: brainy Tested by: murf The specialized reset routine is tromping on the flags field of the CDR. I made a change to not reset the DISABLED bit. This should get rid of this problem. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@139104 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-19Merged revisions 138815 via svnmerge from murf1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines These changes are in regards to bug 13249, where users are being surprised by the changes made to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x installation where a "make samples" was executed, or where they hand-edited the asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher). (this commit does not totally solve 13249, at least not yet) The change involves issueing a single warning while the AEL file is loading, if: 1. app_set is present in the config file, and set to 1.6 or higher. 2. there are double quotes in an assignment statement (eg x = "hi there";) 3. the warning was not already issued. The standalone app, aelparse, does not (yet) issue this warning. I'd have to have it read in the asterisk.conf file, and that's a bit of hassle. I'll add it if users request it, tho. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@138846 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-17Merged revisions 138482 via svnmerge from seanbright1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 lines Move Uniqueid to the end of the event for those that rely on the position of the name/value pairs, pointed out by snuffy-home on #asterisk-commits. For those of you who rely on the position of name/value pairs in manager events... stop... that is why associative arrays were invented. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@138483 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-17Merged revisions 138479 via svnmerge from seanbright1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 lines Add Uniqueid header to ParkedCall manager event. (closes issue #13323) Reported by: srt Patches: 13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@138480 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-17Merged revisions 138476 via svnmerge from seanbright1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 lines Add missing colons to RTCPReceived and RTCPSent manager events. (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@138477 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15Merged revisions 138206 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines Remove deprecated syntax from sample config file (closes issue #13314) Reported by: kue ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@138207 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15Merged revisions 138028 via svnmerge from russell1-9/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) | 17 lines Merged revisions 138027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines Ensure that when a hangup occurs in autoservice, that a hangup frame gets properly deferred to be read from the channel owner when it gets taken out of autoservice. (closes issue #12874) Reported by: dimas Patches: v1-12874.patch uploaded by dimas (license 88) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@138029 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136660 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears once for every bridged call ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136663 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136635 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 lines Don't allow Answer() to accept a negative argument. Negative argument means an infinite delay and we don't want that. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136636 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136633 via svnmerge from mmichelson1-3/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 lines Fix a calculation error I had made in the poll. The poll would reset to 500 ms every time a non-voice frame was received. The total time we poll should be 500 ms, so now we save the amount of time left after the poll returned and use that as our argument for the next call to poll ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136634 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136631 via svnmerge from mmichelson1-23/+33
https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 lines Scrap the 500 ms delay when Asterisk auto-answers a channel. Instead, poll the channel until receiving a voice frame. The cap on this poll is 500 ms. The optional delay is still allowable in the Answer() application, but the delay has been moved back to its original position, after the call to the channel's answer callback. The poll for the voice frame will not happen if a delay is specified when calling Answer(). (closes issue #12708) Reported by: kactus ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136632 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merged revisions 136063 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136064 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merged revisions 135950 via svnmerge from tilghman1-10/+13
https://origsvn.digium.com/svn/asterisk/trunk ................ r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines Fix a longstanding bug in channel walking logic, and fix the explanation to make sense. (Closes issue #13124) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135951 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merged revisions 135938 via svnmerge from tilghman1-1/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) | 4 lines Since powerof() can return an error condition, it's foolhardy not to detect and deal with that condition. (Related to issue #13240) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135942 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merged revisions 135851 via svnmerge from mmichelson3-9/+55
https://origsvn.digium.com/svn/asterisk/trunk ................ r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines Merged revisions 135841,135847,135850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines Remove properties that should not be here ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135852 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Merged revisions 135821 via svnmerge from murf3-12/+45
https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines Merged revisions 135799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135822 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Merged revisions 135598 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug 2008) | 9 lines Merged revisions 135597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line Use PATH_MAX for filenames ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135599 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-02Merged revisions 135265 via svnmerge from murf1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines (closes issue #13202) Reported by: falves11 Tested by: murf falves11 == The changes I introduce here seem to clear up the problem for me. However, if they do not for you, please reopen this bug, and we'll keep digging. The root of this problem seems to be a subtle memory corruption introduced when creating an extension with an empty extension name. While valgrind cannot detect it outside of DEBUG_MALLOC mode, when compiled with DEBUG_MALLOC, this is certain death. The code in main/features.c is a puzzle to me. On the initial module load, the code is attempting to add the parking extension before the features.conf file has even been opened! I just wrapped the offending call with an if() that will not try to add the extension if the extension name is empty. THis seems to solve the corruption, and let the "memory show allocations" work as one would expect. But, really, adding an extension with an empty name is a seriously bad thing to allow, as it will mess up all the pattern matching algorithms, etc. So, I added a statement to the add_extension2 code to return a -1 if this is attempted. in 1.6.0, the changes to only main/pbx.c were applicable, as apparently the code added to main/features by jpeeler were not included in 1.6.0. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135266 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-31Merged revisions 135016 via svnmerge from kpfleming1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul 2008) | 11 lines Merged revisions 134983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines accomodate users who seem to lack a sense of humor :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135034 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-31Merged revisions 134980 via svnmerge from tilghman1-5/+13
https://origsvn.digium.com/svn/asterisk/trunk ................ r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 lines Specify codecs in callfiles and manager, to allow video calls to be set up from callfiles and AMI. (closes issue #9531) Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134981 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-31Merged revisions 134922 via svnmerge from murf1-4/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | 63 lines Merged revisions 134883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines (closes issue #11849) Reported by: greyvoip Tested by: murf OK, a few days of debugging, a bunch of instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook pages of notes later, I have made the small tweek necc. to get the start time right on the second CDR when: A Calls B B answ. A hits Xfer button on sip phone, A dials C and hits the OK button, A hangs up C answers ringing phone B and C converse B and/or C hangs up But does not harm the scenario where: A Calls B B answ. B hits xfer button on sip phone, B dials C and hits the OK button, B hangs up C answers ringing phone A and C converse A and/or C hangs up The difference in start times on the second CDR is because of a Masquerade on the B channel when the xfer number is sent. It ends up replacing the CDR on the B channel with a duplicate, which ends up getting tossed out. We keep a pointer to the first CDR, and update *that* after the bridge closes. But, only if the CDR has changed. I hope this change is specific enough not to muck up any current CDR-based apps. In my defence, I assert that the previous information was wrong, and this change fixes it, and possibly other similar scenarios. I wonder if I should be doing the same thing for the channel, as I did for the peer, but I can't think of a scenario this might affect. I leave it, then, as an exersize for the users, to find the scenario where the chan's CDR changes and loses the proper start time. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134923 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Merged revisions 134703 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines Oops, wrong define ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134706 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Merged revisions 134476 via svnmerge from mmichelson1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a spot where a function could return without bringing a channel out of autoservice. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134477 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28Merged revisions 134125 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 lines This commit compensates for buggy poll(2) implementations. Asterisk has, for a long time, had its own implementation of poll(2) which just used the input arguments to call select(2). In 1.4, this internal implementation was used for Darwin systems. This was removed in Asterisk trunk at some point, but it seems as though this was not the right move to make. On Mac OS X, it appears as though the poll used to gather CLI input does not respond properly when connecting via a remote Asterisk console. Reverting to the use of Asterisk's poll fixed the issue. Also, there is now an option for the configure script, --enable-internal-poll, which will allow for anyone to use Asterisk's internal poll implementation in case they suspect that their system's poll implementation is buggy. closes issue #11928) Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded by putnopvut (license 60) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134126 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28Merged revisions 134086 via svnmerge from kpfleming4-35/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 lines remove remaining Zaptel references in various places ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134087 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28Merged revisions 134050 via svnmerge from mmichelson1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines merging the zap_and_dahdi_trunk branch up to trunk ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@134052 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-26Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, ↵russell1-7/+1
trunk, and 1.6.0. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133982 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-26Merged revisions 133945-133946 via svnmerge from russell1-5/+16
https://origsvn.digium.com/svn/asterisk/trunk ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 Jul 2008) | 6 lines ast_device_state() gets called in two different ways. The first way is when called from elsewhere in Asterisk to find the current state of a device. In that case, we want to use the cached value if it exists. The other way is when processing a device state change. In that case, we do not want to check the cache because returning the last known state is counter productive. ........ r133946 | russell | 2008-07-26 10:16:20 -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache argument ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133947 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Include the http_decode function from trunkbkruse1-3/+20
to replace the + with a space. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133804 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Merged revisions 133665 via svnmerge from tilghman2-36/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) | 16 lines Merged revisions 133649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133671 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-24Merged revisions 133448 via svnmerge from mmichelson1-7/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 lines Print the correct PID in log messages. Prior to this commit, only the logger thread's PID would be printed. (closes issue #13150) Reported by: atis Patches: log_pid.diff uploaded by putnopvut (license 60) Tested by: eliel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133449 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23Merged revisions 133299 via svnmerge from murf1-21/+37
https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines (closes issue #13144) Reported by: murf Tested by: murf For: J. Geis The 'data' field in the ast_exten struct was being 'moved' from the current dialplan to the replacement dialplan. This was not good, as the current dialplan could have problems in the time between the change and when the new dialplan is swapped in. So, I modified the merge_and_delete code to strdup the 'data' field (the args to the app call), and then it's freed as normal. I improved a few messages; I added code to limit the number of calls to the context_merge_incls_swits_igps_other_registrars() to one per context. I don't think having it called multiple times per context was doing anything bad, but it was inefficient. I hope this fixes the problems Mr. Geiss was noting in asterisk-users, see http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133300 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23Merged revisions 133171 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines Merged revisions 133169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@133172 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23Merged revisions 132964 via svnmerge from kpfleming1-11/+42
https://origsvn.digium.com/svn/asterisk/trunk ................ r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul 2008) | 10 lines Merged revisions 132872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132965 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Merged revisions 132511 via svnmerge from tilghman1-0/+31
https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines (Step 2 of 2) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132513 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Merged revisions 132510 via svnmerge from tilghman4-1/+400
https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines Optionally build integer-based routines for FSK tone decoding (but default to the more accurate float-based routines). (Closes issue #11679) (Step 1 of 2) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132512 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Merged revisions 132390 via svnmerge from russell22-10267/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132391 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-18Merged revisions 132242 via svnmerge from bbryant1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines Fixes problem where manager users loaded from users.conf would be removed early (before the routine to load the configuration was finished) because a variable wasn't initialized. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132245 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-18Merged revisions 132113 via svnmerge from tilghman1-19/+75
https://origsvn.digium.com/svn/asterisk/trunk ................ r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) | 6 lines Fix for Taiwanese number syntax (closes issue #12319) Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch uploaded by CharlesWang (license 444) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132114 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-18Merged revisions 132109 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) | 6 lines Textual clarification (closes issue #13106) Reported by: flefoll Patches: config.c.br14.120173.patch-unknown-directive uploaded by flefoll (license 244) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132110 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-18Merged revisions 132050 via svnmerge from bbryant1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and change cdr_radius.c to use the same keyword as the other files (patch by eliel). (closes issue #13104) Reported by: eliel Patches: revision.patch uploaded by eliel (license 64) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132051 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-18Merged revisions 131989 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) | 2 lines Oops ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@131990 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-18Merged revisions 131986 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) | 2 lines Preserve ABI compatibility with last change ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@131987 f38db490-d61c-443f-a65b-d21fe96a405b
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