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2009-03-18Merged revisions 182847 via svnmerge from russell4-20/+21
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@182946 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Merged revisions 182525 via svnmerge from kpfleming1-1/+50
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/1.6.1@182527 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181135 via svnmerge from jpeeler2-14/+39
https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@181199 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Merged revisions 180719 via svnmerge from jpeeler30-418/+774
https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version <ver number> <description of changes> and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180740 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180373 via svnmerge from kpfleming1-0/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180378 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 180032 via svnmerge from dvossel1-0/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180080 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Oops, last merge broke 1.6.1 branchtilghman1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177764 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Merged revisions 177664 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177761 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Merged revisions 177732 via svnmerge from tilghman1-30/+24
https://origsvn.digium.com/svn/asterisk/trunk ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177760 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Fixes issue with undefined audio codecs in chan_iax2dvossel1-0/+2
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177700 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-19Merged revisions 177387 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines Fix another merge error from 176708 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177389 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177098 via svnmerge from tilghman1-3/+59
https://origsvn.digium.com/svn/asterisk/trunk ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600 (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines Document the return value of the update method (as requested on -dev list) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177100 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 176904 via svnmerge from russell1-0/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009) | 2 lines Add example code for a heap traversal. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176906 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176708 via svnmerge from jpeeler1-0/+2
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/1.6.1@176711 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176697 via svnmerge from mmichelson1-3/+11
https://origsvn.digium.com/svn/asterisk/trunk ........ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in frame.h ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176699 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176666 via svnmerge from russell1-13/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176675 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176632 via svnmerge from russell1-0/+218
https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176634 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176255 via svnmerge from kpfleming1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176259 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 175952 via svnmerge from mvanbaak1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176023 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175882 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines Make ast_sched_report() and ast_sched_dump() thread safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175890 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175829 via svnmerge from russell1-3/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175831 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175121 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175123 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-11Merged revisions 174945 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@174947 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173500 via svnmerge from jpeeler1-0/+1
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/1.6.1@173505 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Merged revisions 172580 via svnmerge from twilson1-11/+0
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/1.6.1@172636 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Merged revisions 172598 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@172609 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 172063 via svnmerge from murf1-0/+4
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/1.6.1@172067 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-25Merged revisions 170943 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170945 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-22Merged revisions 169944 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r169944 | tilghman | 2009-01-21 18:44:52 -0600 (Wed, 21 Jan 2009) | 16 lines Merged revisions 169943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@169946 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-17Merged revisions 169080 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r169080 | twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines Fix qualify for TCP peer (closes issue #14192) Reported by: pabelanger Patches: asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@169082 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-16Merged revisions 168832 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168836 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Merged revisions 168562 via svnmerge from russell2-13/+13
https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168565 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-24Merged revisions 166665 via svnmerge from murf2-8/+0
https://origsvn.digium.com/svn/asterisk/trunk This merged from trunk with no conflicts. I tested mostly the 'tired' cases, and for the most part ignored the tests for reconnecting and dialing in to fetch a parked call, after the first case. ................ 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/1.6.1@166730 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-23Merged revisions 166696 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines Allow semicolons and extended characters in user-specified SIP headers. (closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166698 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-22Merged revisions 166282 via svnmerge from russell1-0/+19
https://origsvn.digium.com/svn/asterisk/trunk ........ r166282 | russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166284 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Merged revisions 166092,166095 via svnmerge from mmichelson1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ ........ r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines Remove the verbatim tag from the author line I could have sworn I already did that before, though... ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166098 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165723 via svnmerge from russell1-0/+31
https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165728 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Merged revisions 164737 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) | 22 lines Merged revisions 164736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164739 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164423 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r164423 | mmichelson | 2008-12-15 13:53:29 -0600 (Mon, 15 Dec 2008) | 11 lines Merged revisions 164422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines Add the deadlock note to ast_spawn_extension as well ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164425 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164419 via svnmerge from mmichelson2-0/+25
https://origsvn.digium.com/svn/asterisk/trunk ................ r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec 2008) | 12 lines Merged revisions 164416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164421 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164257 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 | file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164266 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-12Merged revisions 163449 via svnmerge from russell1-1/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163515 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162488 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r162488 | kpfleming | 2008-12-09 17:41:02 -0600 (Tue, 09 Dec 2008) | 1 line it does help if the compiler attribute syntax is correct ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162490 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162414 via svnmerge from russell1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) | 16 lines Merged revisions 162413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162416 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162079 via svnmerge from murf1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) | 53 lines Merged revisions 162013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162082 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Merged revisions 161427 via svnmerge from seanbright1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r161427 | seanbright | 2008-12-05 16:08:43 -0500 (Fri, 05 Dec 2008) | 22 lines Merged revisions 161426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@161429 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-02Merged revisions 160208 via svnmerge from tilghman1-4/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@160234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-29Merged revisions 159818 via svnmerge from kpfleming18-62/+59
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/1.6.1@159851 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-25Merged revisions 159050 via svnmerge from tilghman1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r159050 | tilghman | 2008-11-24 23:02:11 -0600 (Mon, 24 Nov 2008) | 10 lines Merged revisions 159025 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@159053 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-24Merged revisions 158876 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r158876 | mnicholson | 2008-11-24 15:56:22 -0600 (Mon, 24 Nov 2008) | 7 lines Added EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes issue #13873) Reported by: fnordian Patches: ami_agievent.patch uploaded by fnordian (license 110) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@158894 f38db490-d61c-443f-a65b-d21fe96a405b
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