[svn-commits] lmadsen: tag 1.6.1.0-rc2 r180302 - /tags/1.6.1.0-rc2/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Mar 4 18:09:53 CST 2009
Author: lmadsen
Date: Wed Mar 4 18:09:49 2009
New Revision: 180302
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=180302
Log:
Importing files for 1.6.1.0-rc2 release
Added:
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tags/1.6.1.0-rc2/.version (with props)
tags/1.6.1.0-rc2/ChangeLog (with props)
Added: tags/1.6.1.0-rc2/.lastclean
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--- tags/1.6.1.0-rc2/ChangeLog (added)
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+2009-03-04 Leif Madsen <lmadsen at digium.com>
+
+ * Released Asterisk 1.6.1.0-rc2
+
+2009-03-04 21:09 +0000 [r180263] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 |
+ russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines
+ Resolve object matching issues related to the removal of the
+ sip_user object. Previously, chan_sip had both sip_peer and
+ sip_user objects in memory. A patch went in to remove sip_user to
+ simplify the code, since everything could be done with just
+ sip_peer. This patch resolves some regressions found that were
+ introduced by those changes. This code comes from
+ svn/asterisk/team/group/sip-object-matching/. Here is a list of
+ the changes that have been made: 1) When doing a match by name
+ with the find_peer() function, make it much easier to specify
+ which objects should be matched by having a parameter that
+ specifies exactly which object types should be considered. Also,
+ update find_by_name() to handle this parameter. Finally, update
+ all code to use the new option values. 2) When looking up an
+ object for an outbound request by name, consider peers only.
+ (create_addr()) 3) Only match peers on an incoming registration
+ request. 4) When doing authentication (except for SUBSCRIBE),
+ look up users by name, instead of all objects by name. 5) When
+ doing authentication (except for SUBSCRIBE), after looking for a
+ user by name, look for a peer by IP address, instead of all
+ objects by IP address. 6) When handling the SIP qualify CLI
+ command or manager action, look for a peer by name, instead of
+ any object by name. 7) When handling the SIP unregister CLI
+ command, look for a peer by name, instead of any object by name.
+ 9) In sip_do_debug_peer(), search for a peer by name, instead of
+ any object by name. 9) When handling the SIPPEER() dialplan
+ function, search for a peer by name, instead of any object by
+ name. 10) In the following session timer related functions,
+ st_get_se(), st_get_refresher(), and st_get_mode(), when looking
+ for an object for a given sip_pvt using pvt->peername, look for a
+ peer by name, instead of any object by name. 11) Fix build_peer()
+ to properly handle the case where separate type=peer and
+ type=user entries were specified in sip.conf. (closes issue
+ #14505) Reported by: lmadsen Review:
+ http://reviewboard.digium.com/r/172/ ........
+
+2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp <jcolp at digium.com>
+
+ * /, main/callerid.c: Merged revisions 180195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) |
+ 11 lines 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) ........ ................
+
+ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 |
+ file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
+ Remove duplicate 'k' and 'K' Dial options. (closes issue #14601)
+ Reported by: alecdavis Patches: app_dial.optionk.diff.txt
+ uploaded by alecdavis (license 585) ........
+
+2009-03-03 23:39 +0000 [r180080] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, include/asterisk/app.h, apps/app_read.c, /,
+ main/app.c: Merged revisions 180032 via svnmerge from
+ 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/ ........
+
+2009-03-03 23:31 +0000 [r180077] Steve Murphy <murf at digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
+ utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y,
+ main/ast_expr2f.c: Merged revisions 179973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) |
+ 33 lines 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. ........
+ ................
+
+2009-03-03 22:49 +0000 [r179939-180009] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
+ 180007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar
+ 2009) | 22 lines 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 ........ ................
+
+ * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged
+ revisions 179937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 |
+ mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20
+ lines 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/
+ ........
+
+2009-03-03 20:09 +0000 [r179905] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 179903 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar
+ 2009) | 1 line fix a leaked channel lock (and future deadlock)
+ when we try to pick up our own channel ........
+
+2009-03-03 18:30 +0000 [r179843] Joshua Colp <jcolp at digium.com>
+
+ * /, main/features.c: Merged revisions 179841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) |
+ 16 lines Merged revisions 179840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
+ lines Do not assume that the bridge_cdr is still attached to the
+ channel when the 'h' exten is finished executing. It is possible
+ for a masquerade operation to occur when the 'h' exten is
+ operating. This operation moves the CDR records around causing
+ the bridge_cdr to no longer exist on the channel where it is
+ expected to. We can not safely modify it afterwards because of
+ this, so don't even try. (closes issue #14564) Reported by: meric
+ ........ ................
+
+2009-03-03 16:48 +0000 [r179744] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 179742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009)
+ | 14 lines 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. ........
+ ................
+
+2009-03-03 14:41 +0000 [r179674] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c, /: Merged revisions 179672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) |
+ 10 lines 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. ........ ................
+
+2009-03-03 13:56 +0000 [r179611] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 179609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009)
+ | 17 lines 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. ........ ................
+
+2009-03-03 00:04 +0000 [r179539] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 179537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009)
+ | 21 lines 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 ........ ................
+
+2009-03-02 23:39 +0000 [r179535] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009)
+ | 48 lines 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. :-) ........
+ ................
+
+2009-03-02 23:12 +0000 [r179471] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/app.c: Merged revisions 179469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009)
+ | 17 lines 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 ........
+ ................
+
+2009-03-02 23:04 +0000 [r179464] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 179462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009)
+ | 16 lines 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) ........ ................
+
+2009-03-02 20:18 +0000 [r179407] Jason Parker <jparker at digium.com>
+
+ * /, main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: Merged revisions
+ 179396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) |
+ 9 lines 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 ........
+ ................
+
+2009-03-02 17:19 +0000 [r179362] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009)
+ | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not
+ loaded) ........
+
+2009-03-02 14:14 +0000 [r179293] Joshua Colp <jcolp at digium.com>
+
+ * /, main/audiohook.c: Merged revisions 179291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 |
+ file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines 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) ........
+
+2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar
+ 2009) | 5 lines Swap reversed timevals. This was pointed out by
+ ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........
+
+ * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 |
+ mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18
+ lines 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 ........
+
+2009-02-27 21:48 +0000 [r179166] Russell Bryant <russell at digium.com>
+
+ * configs/ais.conf.sample, res/res_ais.c, /,
+ doc/distributed_devstate.txt: Merged revisions 179164 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27
+ Feb 2009) | 2 lines Mark res_ais as experimental, as the binary
+ event format is subject to change. ........
+
+2009-02-27 21:34 +0000 [r179163] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009)
+ | 3 lines If config file is blank, don't load module. (Closes
+ issue #14563) ........
+
+2009-02-27 21:25 +0000 [r179160] Russell Bryant <russell at digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 |
+ russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
+ Add a note about the ordering of entries in sip.conf in 1.6.1.
+ ........
+
+2009-02-27 19:06 +0000 [r179059] Jason Parker <jparker at digium.com>
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 179057 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb
+ 2009) | 8 lines 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) ........
+
+2009-02-27 03:56 +0000 [r178988] Steve Murphy <murf at digium.com>
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 178986 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) |
+ 26 lines Merged revisions 178956 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
+ case, it's just a matter of reducing the default timeouts from
+ 2000 to 1000 msec, as the max def feature digit timeout is no
+ longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
+ -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
+ feature digit timeout to 1000 ms from the previous default of
+ 500. As per bug 14515, a dev discussion arrived at a "mediated
+ concensus" of a default feature digit timeout of 1.0 sec. Some
+ voted for 1300; ctooley thought 1500 for distracted phone users
+ in phone booths; kpfleming put his foot down at 1.0 sec. Users
+ who found the previous default max delay of 250 msec perfect, are
+ welcome to override the new default. Notice that I said that 250
+ msec was the default; wait a minute, you might say, the config
+ file said it was 500 msec!; well, because of the bug fix for
+ 14515, we found that 500 msec was actually enforcing a max of
+ 250. The bug fix would restore 500 msec, but we felt even that
+ was a bit tight for most users... 2000 msec was pushed earlier by
+ mmichelson, so that reduces to 1000 msec after the bug fix.
+ Enjoy! ........ ................
+
+2009-02-26 17:50 +0000 [r178875] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009)
+ | 6 lines IAX2 prune realtime, minor tweak to last fix A return
+ statement was missing which caused unexpected cli output. issue
+ #14479 ........
+
+2009-02-26 17:38 +0000 [r178869] Steve Murphy <murf at digium.com>
+
+ * /, main/features.c: Merged revisions 178828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) |
+ 34 lines Merged revisions 178804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
+ 28 lines This patch prevents the feature detection timeout from
+ being cut in half. Because the ast_channel_bridge() call will
+ return 0 and pass a frame pointer for both DTMF_BEGIN and
+ DTMF_END, the feature_timer field in hte config struct is getting
+ decremented twice, which effectively cuts the digittimeout in
+ half. I added conditions to the if statement to only let DTMF_END
+ frames to flow thru, which solved the problem. Also, when the
+ frame pointer is null, let control flow thru-- this usually
+ happens on timeouts. I added a comment to the code to explain
+ what's going on and why. Many thanks to sodom for reporting this
+ problem. Personnally, it always seemed like something was wrong
+ with the featuredigittimeout, but I never could quite decide
+ what... and was too busy to investigate. This bug forced the
+ issue, and now we know. Sodom had other issues in 14515, but I
+ couldn't reproduce them. If he still has problems, and wants to
+ get them solved, he is welcome to reopen 14515. (closes issue
+ #14515) Reported by: sodom Patches: 14515.patch uploaded by murf
+ (license 17) Tested by: murf, sodom ........ ................
+
+2009-02-26 16:44 +0000 [r178803] Joshua Colp <jcolp at digium.com>
+
+ * /, main/file.c: Merged revisions 178801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 |
+ file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix
+ an issue where the timer for file playback would not be stopped
+ if DAHDI was not installed. (closes issue #14541) Reported by:
+ grant ........
+
+2009-02-26 16:07 +0000 [r178769] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009)
+ | 8 lines 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/ ........
+
+2009-02-25 12:46 +0000 [r178511] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 178509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009)
+ | 10 lines Merged revisions 178508 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
+ | 2 lines Update the copyright year for the main page of the
+ doxygen documentation. ........ ................
+
+2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 178446 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600
+ (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
+ | 5 lines Add section about the #exec command in configuration
+ files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
+ with additional notes by tilghman (license 14) ........
+ ................
+
+ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 |
+ tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
+ Apparently, a void cast doesn't override warn_unused_result.
+ ........
+
+2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant <russell at digium.com>
+
+ * Makefile: revert accidental Makefile change.
+
+ * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r178374 | russell | 2009-02-24 14:39:57 -0600
+ (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
+ | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
+ to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+ russell ........ ................
+
+2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 178375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 |
+ tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
+ The 3 possible errors with pipe(2) are all impossible in this
+ situation. ........
+
+ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24
+ Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of
+ depending upon the astcanary process being inherited by init.
+ ........
+
+ * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 |
+ tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines
+ Cause astcanary to exit if Asterisk exits abnormally and doesn't
+ kill astcanary. Also, add some documentation supporting the use
+ of astcanary. (closes issue #14538) Reported by: KNK Patches:
+ asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
+ ........
+
+2009-02-24 15:22 +0000 [r178232] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) |
+ 16 lines Merged revisions 178205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
+ lines Skip check for extension when subscribing for MWI. Since
+ the remote side is not actually subscribing to a specific
+ extension when subscribing for MWI just skip the check to see if
+ the extension exists. They can't use it to specify the mailbox
+ either since we require configuration of that in sip.conf (closes
+ issue #14531) Reported by: festr ........ ................
+
+2009-02-23 23:22 +0000 [r178172] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c, /: Merged revisions 178142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009)
+ | 22 lines Merged revisions 178141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
+ | 14 lines Fix infinite DTMF when a BEGIN is received without an
+ END. This commit is related to rev 175124 of 1.4 where a previous
+ attempt was made to fix this problem. The problem with the
+ previous patch was that the inserted code needed to go _before_
+ setting the lastrxts to the current timestamp. Because those were
+ the same, the dtmfcount variable was never decremented, and so
+ the END was never sent. In passing, I removed the dtmfsamples
+ variable which was completed unused. I also removed a redundant
+ setting of the lastrxts variable. (closes issue #14460) Reported
+ by: moliveras ........ ................
+
+2009-02-21 16:04 +0000 [r177945] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 |
+ tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines
+ On update, test against the existence of sipregs. ........
+
+2009-02-21 12:51 +0000 [r177851] Michiel van Baak <michiel at vanbaak.info>
+
+ * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 |
+ mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines
+ make chan_sip.c compile on OpenBSD again. ........
+
+2009-02-20 23:05 +0000 [r177789] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, /: Merged revisions 177787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009)
+ | 16 lines Merged revisions 177786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
+ | 9 lines Don't print the CR-NL combination when we aren't
+ outputting to the manager. An embedded CR-NL in a CLI command
+ screws up several AMI parsers that don't expect to see that
+ combination in the middle of output. (Closes issue #14305)
+ Reported by: martins Patch by: tilghman ........ ................
+
+2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 |
+ dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
+ Make app_fax compatible with spandsp-0.0.6pre4 Prior to
+ spandsp-0.0.6pre4 the t30_stats_t structure used a
+ pages_transferred integer to indicate the number of pages
+ transferred (so far) during the fax session. The
+ spandsp-0.0.6pre4 release removed the pages_transferred integer
+ and replaced it with two different integers - pages_tx and
+ pages_rx. This revision uses the new integers for
+ spandsp-0.0.6pre4 while maintaining backwards compatibility for
+ previous spandsp releases. ........
+
+2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch
+
+ * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 177664 via svnmerge from
+ 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 ........
+
+ * include/asterisk/threadstorage.h, /: Merged revisions 177732 via
+ svnmerge from 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. ........ ................
+
+2009-02-20 20:34 +0000 [r177700] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
+ undefined audio codecs in chan_iax2 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
[... 53777 lines stripped ...]
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