asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	eb5bb03b82	Spacing changes only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a66032a14a	Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 \| file \| 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) \| 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3a14487abf	Allow for "magic" pickups to work when we wish to ignore the context When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	15090ba1df	Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	ceea9b1dce	My bad! left check_expr2 in the ALL_UTILS list by mistake. Already done to 1.6.x git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	979eb709ae	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: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	0b01444ab7	Blocked revisions 180010 via svnmerge ........ r180010 \| qwell \| 2009-03-03 17:01:06 -0600 (Tue, 03 Mar 2009) \| 1 line Make sure we still support zapchan in users.conf, in addition to dahdichan. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8970f8caaa	Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 \| mmichelson \| 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) \| 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f47b03877b	Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 \| murf \| 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) \| 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ae786501f1	app_meetme not setting filename and fileformat correctly for realtime When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d8d5e38f65	Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brian Degenhardt	a7092f0acc	fix a leaked channel lock (and future deadlock) when we try to pick up our own channel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	bcf5ecde90	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: https://origsvn.digium.com/svn/asterisk/trunk@179841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7824d71998	Convert pbx_spool to use string fields instead of statically-sized buffers. In tests run after making this conversion, I noticed an approximate 85% reduction in memory usage for call file processing. Review: http://reviewboard.digium.com/r/168/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cfa0d9c0ce	Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 \| russell \| 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) \| 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	f000d5bb0f	Please prefix default values with DEFAULT git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a65727949c	Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 \| file \| 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) \| 3 lines Move where fdno is set to the default value to after the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d9b034a430	Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 \| russell \| 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) \| 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	aa81288bab	Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 \| jpeeler \| 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) \| 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8f6a933e30	Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 \| russell \| 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) \| 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3252cd2e5b	Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 \| tilghman \| 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) \| 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d6652e9760	Fix a reference leak in timerfd_set_rate(). (found during a debugging session with dvossel and mmichelson.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0c0479602e	Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 \| russell \| 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) \| 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	9bb9c64521	Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 \| qwell \| 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) \| 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5120334acc	Backport 1.6.0 fix to trunk (failsafe if db is not loaded) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	775b30307f	Do not try to remove a registration scheduled item if the scheduler context has already been destroyed. (closes issue #14580) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	93749ba001	Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	932cef55fd	Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c9252cbaf0	Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d2c5b0f1de	Mark res_ais as experimental, as the binary event format is subject to change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	31ae1bfa0d	If config file is blank, don't load module. (Closes issue #14563) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a80ef0ca1e	Add a note about the ordering of entries in sip.conf in 1.6.1. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	99ad18c4c5	Add reload support to chan_skinny. Special thanks goes to DEA who had to redo this patch twice because we first put unload/load support in and later redid the way we configure devices and lines. (closes issue #10297) Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA (license 3) With mods by me based on feedback from wedhorn and Russell and seanbright Tested by: DEA, mvanbaak, pj Review: http://reviewboard.digium.com/r/130/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	796ec24ed5	Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fed7b94f0b	Fix downloading SIREN7 and SIREN14 sound packages. In passing, also fix downloading SLIN16 extra sound packages. (closes issue #14565) Reported by: jtodd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	ec6101595e	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: https://origsvn.digium.com/svn/asterisk/trunk@178986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	63561aea00	Sound confirmation of call pickup success. (closes issue #13826) Reported by: azielke Patches: pickupsound2-trunk.patch uploaded by azielke (license 548) __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3d0aac6cd8	IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	22ea970494	These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	5421f204bc	Blocked revisions 178838 via svnmerge ........ r178838 \| dvossel \| 2009-02-26 11:24:02 -0600 (Thu, 26 Feb 2009) \| 9 lines IAX2 prune realtime fix Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	fe216b2f9d	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: https://origsvn.digium.com/svn/asterisk/trunk@178828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5f7f4a0c84	Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a0ef434095	IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3c342501e3	Ensure there is a valid tone part before trying to play tones. (closes issue #14558) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	775ffb66d0	Clarifications on the different models and reference to further docs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	28eedaaf9c	another minor commit to test post-commit script changes (now testing post-revprop-change as well, third try) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	231823dafe	minor commit to test post-commit script changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	37c1aecb33	Blocked revisions 178640 via svnmerge ........ r178640 \| murf \| 2009-02-25 14:00:50 -0700 (Wed, 25 Feb 2009) \| 17 lines This patch completes the fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) 8-bit transparent While I was updating ast_expr2.fl, I missed one rule that would allow 8-bit chars to be caught in tokens; and in so doing, it absorbs the ${ sequence and messes up the checking of raw exprs by AEL. Trunk already has these changes. (closes issue #14543) Reported by: klaus3000 Patches: patch.14543 uploaded by murf (license 17) Tested by: murf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	baf144c655	Picky, picky buildbots git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4ac2fd4cde	Use notification when timezone files change and re-scan then. (closes issue #14300) Reported by: jamessan Patches: 20090127__bug14300.diff.txt uploaded by tilghman (license 14) 20090224__bug14300.diff uploaded by jamessan (license 246) Tested by: jamessan Review: http://reviewboard.digium.com/r/136/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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